As of February 1, 2012, Genesys is no longer an affiliate of Alcatel-Lucent; any indication of such affiliation within Genesys products or packaging is no longer applicable. Please see the Genesys website at http://www.genesys.com for more details.
This release note applies to all 8.1 releases of SIP Server.
Genesys follows applicable third-party redistribution policies to the extent that Genesys solutions utilize third-party functionality. For additional information on third-party software used in this product, see the Legal Notices for SIP Server. Please contact your Genesys Customer Care representative if you have any questions.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.38. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
true
, after agent logins on the second node, the original agent session node distributes EventAgentLogout.
The new option, send-agent-logout-on-dr, has the default value of false
. It can be set to true
on SIP Server Application level and the change takes effect immediately.
Note: This option is reserved by Genesys Engineering. Use it only when requested by Genesys Customer Care.
(SIP-28861)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.38. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
ipc-maximum-send-timeout
, is introduced to set a timeout to compensate in case of a lengthy freeze (due to CPU starvation) of SIP Server in cluster mode to avoid internal connection disruptions.
Note: This option is reserved by Genesys Engineering. Use it only when requested by Genesys Customer Care.
(SIP-28417)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.38. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
CALL_OBJECT
class. Now, the ISCC::CALL_OBJECT
object is deleted after end of the call. (SIP-28417)
true
, and record-consult-calls = false-extended
were set. (SIP-28807)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
false-extended
, as follows:TServer
false
true, false, false-extended
false
, SIP Server does not allow recording for consultation calls even if one or more of the participating DNs are set for call recording (record option on the DN is set to true
, or the DN is the party specified in the record extension of a TRouteCall request). Not including multisite (ISCC) calls.false-extended
, SIP Server does not allow recording for consultation calls for all type of calls.false-extended
, the call will not be recorded even if it was delivered by ISCC.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
In SIP Cluster mode, the internal disconnect and reconnect between Session Controller (SC) and the Interaction Proxy (IProxy) are handled properly now.
Previously, IPProxy stopped sending TEvents to ICONS after a reconnection. (SIP-28767)
In SIP Cluster mode, when the option enable-strict-location-match is set to true
, SIP Server selects the correct msml service by geo-location, even if there are multiple services with overlapping geo-locations.
Previously, it could select a service with geo-location with a partial match. For example, for a call with geo-location=X
, it could select a service with geo-location=TESTX
. (SIP-28758)
In a multisite call monitoring scenario, SIP Server now correctly handles deregistration of devices, monitored from an aggregated desktop client.
Previously, SIP Server would cancel remote supervision session, initiated from one device, when deregistering another device from the same aggregated client. (SIP-28739)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
This release contains the following corrections and modifications:
SIP Server will not terminate unexpectedly in scenarios, where DN of type Extension configured to request REGISTER
authorization, but such an authorization is never provided by SIP End Point, and DN was disabled in CME.
Previously, SIP Server might have experienced failure in such scenarios. (SIP-28721)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
TServer
SSLv23
SSLv23, SSLv3, TLSv1, TLSv11, TLSv12, TLSv13
SSLv23
- The highest TLS protocol version supported by Genesys Security Pack 8.5.1. Currently, it is TLS version 1.3.SSLv3
- SSL version 3.0TLSv1
- TLS version 1.0TLSv11
- TLS version 1.1TLSv12
- TLS version 1.2TLSv13
- TLS version 1.3SSLv23
(default), SIP Server uses the highest TLS version supported by Genesys Security Pack 8.5.1. Currently, it is TLS 1.3.This release contains the following corrections and modifications:
SIP Server no longer terminates unexpectedly on getting a TCP transport exception while waiting for a REFER message response. (SIP-28704)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
In case of multi-site supervision in SIP Cluster environment, if a TMonitorNextCall request to monitor remote agent fails by timeout, SIP Server now processes further monitoring requests issued by the supervisor successfully. Previously, SIP Server maintained the subscription internally when request failed by timeout, leading to rejects for further supervision requests for this supervisor DN. (SIP-28602)
When SIP Trunk or softswitch required authentication of session refresh re-INVITE, SIP Server now correctly processes following T-Library requests (like hold or retrieve). Previously, refresh re-INVITE transaction with authentication was not completed correctly, and SIP Server could be set into a state where some requests were ignored. (SIP-28689)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
SIP Server no longer terminates unexpectedly in a race condition scenario, when Agent does a RequestCompleteConference
and releases the call before the conference is established, and then the customer leg is transferred to another Routing Point and SIP Server receives a TApplyTreatment
request on an already released call. (SIP-28586)
SIP Server no longer terminates unexpectedly while handling a REFER
request with an invalid User-to-User
parameter in the Refer-To
header. (SIP-28624)
In SIP Cluster mode, when the cluster-wrap-up-0-from-udata
option is set to true, and WrapUpTime is not defined in UserData nor in the relevant extension during agent login and the wrap-up-time
application option has a non-zero value, SIP server will now correctly put an agent into the ACW state when a business call is released. Previously, in such cases, the wrap-up-time
application level option was not processed correctly. (SIP-28639)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
When an inbound call to an agent DN, where an agent is logged in, and recording and greeting are both scheduled, is hung up, the call will be fully terminated irrespective of the recording or greeting or supervisor stage that the call is in at that point. Previously, the call was left stuck in cases when it was hung up in the middle of recording initialization.(SIP-28587)
A new configuration option, sip-schedule-record-on-hold, is introduced to avoid scheduling a recording for the call hold party if there is any other record-enabled party on the call. If no other record-enabled party is on the call, the recording will be scheduled once the call hold party retrieves the call.
sip-schedule-record-on-hold
Setting: [TServer] section, Application level or DN level
Default Value: true
Valid Values: true, false
Changes Take Effect: On the next call
Setting the option to false
will prevent automatic retrieving of the party on hold when creating a recording leg. The default value is true
and retains the existing behaviour.
(SIP-28571)
Recordings are no longer broken when there are race conditions between session refresh re-INVITEs to members of a conference and the re-INVITE from SIP Endpoint (hold or retrieve operation). Previously, recordings could be broken at times in similar circumstances. (SIP-28575)
SIP Server now correctly escapes the 0x7F (DEL) character in attached user data. Previously, the character was not escaped, which led to GVP not responding to the SIP Server INVITE request containing user data with that character. (SIP-28598)
Supported Operating Systems
New in This Release
Corrections and Modifications
As of October 10, 2023, this release is no longer available. A critical issue was discovered. If you downloaded the software, do not install it. The identified issue will be addressed in the next release.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
Now, SIP Server rejects a MergeCalls
request with an error if AttributeTransferConnID
points to a call that has already been merged or released. Previously, SIP Server might erroneously start to process such a call and as a result the MergeCalls
request did not receive any response. (SIP-28538)
SIP Server can now apply dial-plan rules to a call moved from one Routing Point (RP) or external RP to another RP. To control this capability, a new application option, enable-rp-to-rp-dial-plan, is introduced.
enable-rp-to-rp-dial-plan
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: On the next call
If set to true
, SIP Server applies dial-plan rules to a call moved from one routing point to another. If set to false
, dial-plan rules are not applied and the call is just sent to another routing point. For multi-site ISCC calls, additionally, the enable-iscc-dial-plan option must be set to true
. For regular calls, additionally, the rp-use-dial-plan option must be set to full
or partial
.
Note: Between versions 8.1.103.11 and 8.1.104.68, SIP Server did not apply dial-plan rules for calls that were moved from one routing point to another.
(SIP-28578)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
The complete call is recorded when a call is routed to an agent with record
set to source
in AttributeExtensions of a RequestRouteCall and then routed to another agent with record
set to destination
. Previously, the second part of the call was not recorded in such cases. (SIP-28521)
SIP Server now generates an EventPartyChanged event with the cause as Transferred
in scenarios involving a conference call with external parties and where the external party performs a single-step transfer. Previously, SIP Server generated an EventPartyDeleted event in such scenarios. A new option, report-external-sst, is introduced to control the behaviour of the application in such scenarios.
report-external-sst
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: On the next call
When the option is set to true
, SIP Server correctly reports scenarios involving single-step transfer on external (non-monitored) devices, resulting in consistent reporting. Setting the option to false
ensures backward compatibility and retains the old behaviour.
(SIP-28535)
In a 3-way conference call with recording, when one participant is put on hold and if the recording fails with no possibility of a recovery, the participant is now put back on hold. Previously, the participant was included back into the conference call. (SIP-28542)
When running in SIP Cluster mode, SIP Server now correctly processes a value of 0
for the WrapUpTime key present in UserData. Previously, when the value of the WrapUpTime extension in UserData was 0
, SIP Cluster selected the value configured in the application. A new option, cluster-wrap-up-0-from-udata, is introduced to specify how SIP Cluster behaves in such a scenario. The new option is applicable only for SIP Cluster.
cluster-wrap-up-0-from-udata
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: On the next call
When the option is set to true
, SIP Server in SIP Cluster mode will process a value of 0
for the WrapUpTime key present in UserData the same way as SIP Server does.
(SIP-28554)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.35. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
false
true
, false
If set to true
, SIP Server, while deployed in SIP Cluster mode, will eliminate processing of agent login objects by the tconf library. This allows to speed up initialization of the SIP Cluster node even if the number of configured agent logins exceeds 100,000.
(SIP-28107)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.33. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
The following new configuration option is introduced to enable this enhanced restriction of 3PCC requests:
3pcc-requires-agent-sessionfalse
true
, false
When set to true
, SIP Server only accepts 3PCC requests from a client associated with an agent login session. Requests from other clients are rejected with EventError error code: 118 (Requested service unavailable) 'Access restricted'
. When set to false
, SIP Server does not restrict 3PCC requests based on an agent login association. This option is only applicable for requests with AttributeThisDN of types Extension and ACD Position.
For more information on this, refer to the 3pcc-requires-agent-session configuration option description the Supplement to SIP Server Deployment Guide.
(SIP-28460)
This release includes the following corrections and modifications:
When agent DN and supervisor DN are configured with the dual-dialog-enabled option set to single-dialog-rtp-on-hold
, recording is enabled, and nailed-up connections configured, a caller is no longer disconnected while on hold or right after an agent retrieves the caller from a hold. Previously, the caller was disconnected in certain cases where the supervisor initiated monitoring of an agent when the caller was on hold. (SIP-28446)
Now, when the INVITE on a predictive call is rejected with the error code 481, the CallState
in EventReleased
can be set to a value that is not 0. The exact value of the CallState
can be controlled by the sip-error-conversion option. Previously, SIP Server always generated the CallSate
as 0 for predictive calls that ended with the 481 response. (SIP-28448)
When the dual-dialog-enabled option is set to single-dialog-rtp-on-hold
, an agent can now retrieve a caller from hold after disconnecting from a consult with another agent. Previously, agents were unable to retrieve a caller from hold after disconnecting from a consult with another agent in some cases. (SIP-28472)
In cases where DN of Virtual type is enabled after a SIP Server restart, an EventDNBackInService
message is generated. Previously, an EventDNOutOfService
message was sent right after an EventDNBackInService
, for enabled Virtual devices. (SIP-28500)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.33. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
Now, during startup, SIP Server in a cluster mode suppresses remote ADDP timeout on connection to Config Server while it processes the Agent Login CME objects. This eliminates the possibility of Config Server disconnecting from SIP Server due to an ADDP timeout. (SIP-28402)
SIP Server now handles and responds properly when both UPDATE
and PRACK
requests are sent in a call. Previously, when SIP Server received a PRACK message for the call, while waiting for a response for the sent UPDATE message, SIP Server did not handle this race condition and terminated the call. (SIP-28411)
When the auto-logout-timeout option is changed in SIP Cluster, agents are logged out correctly as per the new value specified in the option. Previously, the elapsed time was compared with the remaining time, that is the difference between the old value and the new value of the option, and as a result, agents were logged out incorrectly. (SIP-28445)
Supervision is established correctly, in cases where after a supervisor initiates a monitoring session, the supervisor DN goes OOS/In-Service. Previously in such cases, supervision was not established correctly. (SIP-28457)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
Now, a DN is set into the Out of Service
state only when a SIP registration expires or a DN explicitly unregisters from SIP Server. Previously, since version 8.1.104.21, SIP Server set a DN into the Out of Service
state when a SIP REGISTER
request was rejected with a 401
response. (SIP-28235)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following corrections and modifications:
Now, a SIP request is not sent more than twice to any configured (or returned by DNS server) SIP Proxy address that is unavailable. Previously, when all configured SIP Proxies were not available, SIP Server may send an unlimited number of SIP requests through the pool of configured SIP Proxies. And the requests continued unless all the proxies were set into the Out Of Service
state by failed OPTIONS
requests. (SIP-28005)
In a 3-way conference with recording, when one participant is put on hold, if the recording fails and is then recovered, the participant is put back on hold. Previously, the participant was included back into the conference audio when the recording was recovered.(SIP-28413)
When running in SIP Cluster mode, and if a client registered on a Smart Proxy, it will now receive events on the Routing Point, created after a SIP Server instance was started. Previously, this was not dynamic and events were recieved only after SIP Server was restarted. (SIP-28420)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following modification:
In DR mode, a non-emergency DN without an active registration is not set to Back-In-Service
on the INVITE initiating a first-party call-control (1pcc) call if the new dr-back-in-service-on-invite option is set to false
.
The following new configuration option is introduced:
dr-back-in-service-on-invitetrue
true
, false
Previously, setting such devices to Back-In-Service
allowed agents to login and set their DN as ready
without having their device ready for voice calls resulting in EventError (Ivalid Called Dn)
responses to any attempt to route calls to such DNs. (SIP-28293)
When an agent's SIP phone is configured to support multiple media like RTP and SRTP, hold/retrieve operations are processed correctly. Previously, in rare cases, agent was not able to perform a hold/retrieve operation more than once on a call. (SIP-28364)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following modification:
Now, SIP Server threads establish connections when TLS is configured for T-Controller, Smart Proxy, and default ports. Before the fix, these connections failed over TLS. (SIP-28329)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release contains the following modification:
When operating in the Disaster Recovery (DR) mode, if a DR peer trunk is out of service and an inbound call arrives at an unregistered DN, the call is now routed correctly in accordance with the dial plan's onnotreg rule. Previously, in such a scenario, the call was rejected. (SIP-28321)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
A nailed-up connection which is dropped with a 4xx
response is now re-established correctly when an agent changes the status from Not Ready to Ready. (SIP-28259)
SIP Server no longer terminates unexpectedly when an INVITE containing the Replaces
header is received on a rejected call (with a 603 Decline or 404 Not Found error message). A 481 Call/Transaction Does Not Exist message is sent as a response to the INVITE to prevent the race condition that previously caused the unexpected termination. (SIP-28301)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When the monitor-internal-calls option is set to false
, a supervisor is now not engaged into an internal call when the call is retrieved from hold. Previously, supervision might be activated for a supervisor subscribed to monitor an agent, when an internal call to that agent is retrieved from hold. (SIP-28257)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
Now, TLib-based clients do not disconnect from SIP Server after a switchover. Previously, this issue occurred intermittently due to a delay during switchover. (SIP-28159)
SIP Server no longer grows in memory while processing a treatment recovery. Previously, some internal objects were not released during the treatment recovery attempt, which led to growing memory usage. (SIP-28201)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now properly updates the new overflow location of the call according to the overflow-location-map configuration option in the following scenario:
msml
(and)
503 Service Unavailable
error (and)
Previously, the new overflow location was not updated in the above scenario and as a result, SIP Server did not send a media or recording request to GVP for outbound calls.
(SIP-28183)
Agent Assist functionality is not enabled for the consult call even if the Consult Agent has an Agent Assist profile set up in options or Agent Assist parameters are sent via a RouteCall
request. This means, SIP Server will not send Agent Assist parameters in the INFO message to GVP, to start recording for a Consult Agent. Previously, Agent Assist parameters, if available for a Consult Agent, were sent by SIP Server to GVP when recording started for the consult call.
Related feature documentation: Passing Extended Recording Metadata to GVP in the Framework 8.1 SIP Server Deployment Guide.
(SIP-28199)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When UserData is mapped from TLib to SIP, SIP Server no longer adds duplicate UserData key-value pairs. Previously, when the userdata-map-filter-mode option was set to block
and the userdata-map-filter option was set to the list of prefixes, some UserData values were duplicated. (SIP-28167)
Now, when the greeting-delay-events option is set to true
, the caller and the agent are reconnected properly after a failed greeting. Previously, when SIP Server received a 603 error response for the next INVITE request to the media leg, the caller and the agent were not reconnected. (SIP-28141)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality.
The following new configuration option is introduced:
retry-after-reliable-transport-error0
0
to 34000
(recommended - 500
to 5000
)The option specifies the number of milliseconds that SIP Server will wait before re-sending a request after a transport failure on re-INVITE. If the option is set to 0
or not configured, SIP Server disconnects the SIP peer after a transport error.
For more information on this, refer to the Re-establishing connection on TCP transport exception topic in the Supplement to SIP Server Deployment Guide.
Check SIP Proxy Release Notes for availability of the Reliable Transport Failure for Deployment with SIP Proxy functionality.
(SIP-27982)
This release includes the following corrections and modifications:
A new confirguration option is introduced to address a rare scenario where a DN does not go out of service during a race condition when use-register-for-service-state is set to true
and authenticate-requests is set to register
.
avoid-registrar-ha-race-condition
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true
, false
Changes Take Effect: Immediately
When set to true
, SIP Server may eliminate a possible race condition related to SIP registration which may happen during SIP Server start-up or HA switchover. This setting may help in environments where during start-up, SIP Server experiences a long time delay (up to 1 second or more) for interthread communication.
Important: Consult with Genesys Customer Care before configuring this option.
(SIP-27734)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
The reuse-tls-conn option is now also supported at the DN level. Previously, this option was supported only at the Application level. (SIP-28030)
In a rare occurrence, when SIP Server in cluster mode switched from a primary instance to a backup instance, an agent was unable to log in to Agent Desktop even after resetting the password. This issue is now resolved. (SIP-28095).
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
Now, Agent 1 is correctly retrieved from a hold and all parties in a conference hear each other in the following scenario:
single-dialog-rtp-on-hold
(and)
Previously, the retrieve failed with an error.
(SIP-28070)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality.
This release includes the following corrections and modifications:
Now, a supervisor is properly released from a call in the following scenario:
true
(and)
Agent
initiates a consultation call (and)
(SIP-27877)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
Now, when SIP Server is working in a multi-threaded mode (sip-link-type option is set to 3
) and the snapshot log option is set to a different directory, all snapshot logs from all threads are written to the specified directory. Previously, only snapshot logs from the main thread were written to the specified directory. (SIP-26794)
When the record-moh option is set to false
, one party in the recorded call is put on hold, and a 480 Temporarily Unavailable
error message is received from GVP after a recording failure for attempted recovery, SIP Server now strips the recording and allows the call to continue. Previously, the call was terminated in such scenarios. (SIP-27949)
If the reuse-tls-conn option is set to false
, SIP Server now correctly does not reuse the existing TLS transport for sending SIP requests. Previously, it was possible to reuse a TLS connection irrespective of the value of the reuse-tls-conn option. (SIP-27964)
When a call is received at a Hunt Group where all the Hunt Group members' DNs are not registered and are configured with voice mailbox, the call is correctly distributed to a default-dn
. Previously, such calls were routed to the first member's voice mailbox. (SIP-27972)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now distributes the EventPartyAdded event with AttributeCallState set to 2
in a multi-site single step conference scenario. Previously, SIP Server distributed the EventPartyAdded event with AttributeCallState set to 0
and as a result, WDE no longer considered the call a conference call and the agent was unable to suspend or delete a party from the conference. (SIP-27618)
partyadded-def-callstate-conf
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: On the next call
If the new option is not present or the value is set to false
, SIP Server continues to send the EventPartyAdded event with AttributeCallState set to 0
. If the option is true
, then AttributeCallState is set to 2
.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In cases where a call is established with 2 media lines (RTP/AVP and RTP/SAVP), and then the RTP/SAVP media line needs to be disabled, SIP Server now correctly sends an UPDATE message with the disabled media line as m=audio 0 RTP/SAVP 0
. Previously, SIP Server sent the media line as m=audio 0 RTP/AVP 0
. (SIP-27912)
When a call is established between two agents, Agent 1 and Agent 2, and Agent 2 makes a Single Step Transfer to an external number, and Agent 1 makes a Single Step Conference to Agent 2, SIP Server now generates the EventPartyAdded
event to indicate a conference. Previously, the EventPartyAdded
event was not generated in such a scenario. (SIP-27928)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In a multisite deployment, when the reuse-sdp-on-reinvite option is set to true
on the outbound Trunk DN, and a call is established between the SIP Server sites, SIP Server now correctly propagates a re-INVITE with no SDP to the other site. Previously, in a similar scenario, such a re-INVITE might not be propagated. (SIP-27939)
While working in SIP Cluster mode, SIP Server now correctly applies ACW after a business call. Previously, in some scenarios where the next business call was made when ACW for the previous business call was still in progress, ACW was not applied consistently. (SIP-27157)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.29. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly connects the call in a scenario where the reuse-sdp-on-reinvite option is true
and the sip-treatment-continuous option is false
in the SIP Server application, when an inbound call with active recording is transferred to a Routing Point using a TSingleStepTransfer request and then the call is routed to a destination. (SIP-27817)
While reconnecting to Configuration Server, SIP Server now correctly repeats attempts to re-read connection parameters of the remote SIP Server from Configuration Server. Previously, after reconnection to Configuration Server, if the attempt to re-read the configuration resulted in an error, the connection between SIP Servers (through ISCC) might not be restored. (SIP-27924, TS-12101)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When a single-step transfer is made to a Routing Point and an EventQueued message is generated at the Routing Point, SIP Server no longer incorrectly adds a duplicate CallUUID key in AttributeUserData. And, when UserData is mapped to SIP headers (the userdata-map-trans-prefix option is configured), SIP Server no longer adds a duplicate X-Genesys-CallUUID header in the re-INVITE message. (SIP-27896)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
Support for remote agent deletion from a conference (TDeleteFromConference) in the mixed SIP Server and T-Server for Cisco Unified Communications Manager (CUCM) multisite environment.
SIP Server supports the following scenarios:
A remote TDeleteFromConference request is a TDeleteFromConference request with a special key-value pair location=remote_tserver_location
in AttributeExtensions.
While processing the TDeleteFromConference request, SIP Server does not verify AttributeOtherDN or the location value. The remote server executes this request only if the location value matches the server location and OtherDN could be found on the call. If the remote server ignores the request, the initiating server responds with Error Code 57 (ErrorTimeout) after the timeout expires.
This feature does not depend on the LCT party (Call Participant Info) functionality.
The new value of hybrid
is added to the existing sip-remote-del-from-conf configuration option. In multisite deployments, when this option is set to true
, SIP Server processes a TDeleteFromConference request to remove a remote party (specified in OtherDN) from a conference.
With a value of hybrid
, SIP Server processes a TDeleteFromConference request even if the remote server is not a SIP Server (for example, T-Server for CUCM). The remote deletion from a conference could be done on the SIP Server peer, on the T-Server for CUCM peer, and any other T-Server that supports the same remote deletion from a conference rules as T-Server for CUCM supports.
(SIP-27781)
This release includes the following corrections and modifications:
While adding, updating, or deleting configuration options in a DN object that is in enabled state and out of service, SIP Server no longer sends EventDNBackInService messages to T-Library clients. Previously, SIP Server sent EventDNBackInService messages when some options were set, updated, or deleted. (SIP-27886)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
The override-domain-from and override-from-on-conf configuration options now work independently. The override-domain-from option controls the domain name of the From header in the URI and the override-from-on-conf option controls the username part of the From header in the URI. Previously, if both options were configured, the value of the override-from-on-conference option canceled the value of the override-domain-from option. (SIP-27833)
The SIP Server's template, TServer_SIPPremise_811.xml, now includes all supported values of the use-data-from configuration option. Previously, the value of active-data-original-call
was missing in the template. (SIP-27787)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When the dual-dialog-enabled option is set to single-dialog-rtp-on-hold
, a call is transferred from Agent 1 to Agent 2, and both agents are placed on hold, Agent 1 is correctly retrieved from hold. Previously, SIP Server incorrectly played music-on-hold after receiving a TRetrieveCall request from Agent 1. (SIP-27773)
SIP Server in SIP Cluster mode clears internal DN ownership information (in the SIP Cluster node) when a corresponding DN is deleted from the Configuration Database or a configured ownership type in the DN contact option is changed, as follows:
Previously, DN ownership information sometimes became stuck even after the DN was deleted and re-created in the Configuration Database. (SIP-27376)
When a trunk DN with the name "dummy" is placed into the out-of-service state without registration, this "dummy" DN no longer conflicts with other active trunks during a service check. Previously, when a "dummy" named DN was out of service, SIP Server rejected calls to and from other active trunks in the environment. (SIP-23832)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
In a SIP Cluster environment with more than three nodes and the Smart Proxy enabled, DNs owned by particular T-Controllers are no longer placed in the out-of-service state after recovery from a complicated split-brain scenario. (SIP-27561)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer sends NOTIFY presence messages to subscribers after their subscription expires. (SIP-27727)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server adds a new Application-level option, reuse-tls-conn, to enable reuse of the existing TLS transport for sending SIP requests. In the environments where SIP requests are rejected if SIP Server reuses the existing TLS transport (based on the incoming connection), set this option to false
. (SIP-27730)
reuse-tls-conn
Setting: [TServer] section, Application level
Default Value: true
Valid Values: true, false
Changes Take Effect: On the next call
Specifies whether SIP Server reuses the existing TLS transport for sending SIP requests. If set to false
, SIP Server opens a new TLS connection to the SIP request destination. If set to true
, SIP Server reuses the existing TLS transport for sending SIP requests.
In a scenario where the clamp-dtmf-allowed option is set to true
, DTMF clamping is applied to the conference, and a call-originating party leaves the call, SIP Server now correctly disables the DTMF clamping. Previously, SIP Server did not disable the DTMF clamping in such scenarios. (SIP-27639)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
SIP Server now supports the criteria for selecting inbound trunks where there are several Trunk DNs with the same contact option value configured. Two new options are introduced: the inbound-trunk-hint-sip-field option at the Application level and the inbound-trunk-hint option at the DN level. When the inbound-trunk-hint-sip-field option is configured, it enables the inbound trunk selection criteria. SIP Server selects a trunk as inbound that has a value of the inbound-trunk-hint option.
inbound-trunk-hint-sip-field
Setting: [TServer] section, Application level
Default Value: No default value
Valid Values: A string
Changes Take Effect: On the next call
Defines which SIP header (for example, X-CarrierID) in a SIP INVITE message SIP Server uses for matching against the value configured in the DN-level inbound-trunk-hint option set on the Trunk DN.
inbound-trunk-hint
Setting: [TServer] section, DN level (Trunk DNs)
Default Value: No default value
Valid Values: A string
Changes Take Effect: On the next call
Dependent option: inbound-trunk-hint-sip-field
Specifies the value of the SIP header that is defined in the Application-level inbound-trunk-hint-sip-field option. SIP Server uses this SIP header value to select the best suitable trunk among other trunks with the same contact option value. The inbound-trunk-hint option applies only if the Application-level inbound-trunk-hint-sip-field is configured.
If the inbound-trunk-hint option is set to an asterisk (*) as a wildcard, SIP Server gives preference to selecting trunks that contain this option as inbound, as compared to trunks that do not have this option configured.
Example:
X-CarrierID
, and Trunk DNs are configured as follows:
address1
, inbound-trunk-hint = carrier1
address1
, inbound-trunk-hint = carrier2
(SIP-27510)
There are no corrections or modifications in this release.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When two calls are merged by a transfer, SIP Server checks whether a beep should be played based on the beep parameter setting in AttributeExtensions of the TMakeCall request. Previously, SIP Server ignored the beep setting in AttributeExtensions of the TMakeCall request and played a beep based on the Trunk Group DN beep option that was set to on
by default. (SIP-27675)
When the preview interaction expires at the same time as a session refresh is generated, SIP Server now completes the session refresh and then processes the preview interaction expiration event. Previously, in this race-condition scenario, SIP Server generated an error message. (SIP-27636)
In a multisite call routing scenario with DTMF clamping enabled (clamp-dtmf-allowed = true
) and divert-on-ringing = false
, SIP Server no longer incorrectly plays the conference music (set in the music-in-conference-file option) to a call which an agent places on hold. (SIP-27610)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.27. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
userdata-map-filter-mode
Setting: [TServer] section, Application level
Default Value: allow
Valid Values: allow, block
Changes Take Effect: Immediately
Specifies whether the patterns, provided in the userdata-map-filter option, are allowed or blocked for mapping the matching T-Library UserData to SIP headers.
block
, and:
allow
, and:
The following existing option is modified to support this functionality:
userdata-map-filter
Setting: [TServer] section, DN level
Default Value: No default value
Valid Values:
Specifies a prefix (or a list of prefixes) that must match the initial characters of the key in the UserData key-value pair. When the initial characters match, then SIP Server either allows or blocks mapping of UserData into SIP messages, based on the setting in the userdata-map-filter-mode option.
For example, if userdata-map-filter=test
and AttributeUserData contains 'test'='value1', 'testlocal'='value2',
and 'generaltest'='value3'
, only key-value pairs 'test'='value1'
and 'testlocal'='value2'
are matched the prefix pattern and considered for mapping. The 'generaltest'='value3'
is ignored because its initial characters do not have the prefix test
.
If this option is not specified, no data will be mapped.
(SIP-27563)
This release includes the following corrections and modifications:
SIP Server can now process postponed configuration events asynchronously while reconnecting to Configuration Server. To enable asynchronous processing of postponed configuration events, set the tconf-enable-async-processing option to true
in the TServer section. See the option description in T-Server Common Part (TSCP) release number 8.5.000.27. (SIP-27679)
When sip-early-dialog-mode is enabled, SIP Server now correctly handles the UPDATE request that SIP Server receives while converting an early dialog to a full dialog. Previously, in this scenario, SIP Server did not process the UPDATE request and disconnected the call. (SIP-27597)
When the send-200-on-clear-call option is set to true
, SIP Server now responds to the CANCEL message by sending a 200 OK message with the initial SDP offer. Previously, SIP Server responded to the CANCEL message by sending a 200 OK message with the default SDP content. (SIP-27517)
When sip-enable-sdp-codec-filter is enabled, SIP Server adds support of the RTP/AVPF codec in the incoming SDP offer. Previously, SIP Server converted the RTP/AVPF codec to RTP/AVP and sent it in the SDP offer. (SIP-27495)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.26. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When use-register-for-service-state is set to true
and authenticate-requests is set to register
, and a client sends a REGISTER request without the Authorization header, SIP Server now correctly places the corresponding DN in the out-of-service status. Previously, SIP Server did not place that DN in out-of-service. (SIP-27595)
When divert-on-ringing is set to false
and one leg of the call on a Routing Point is in the connected state (an agent is connected to Media Server),
and the second leg, while in early dialog state, sends a 183 (Session in Progress) message with the SDP, and then immediately 200 OK with the SDP, SIP Server now correctly generates EventRouteUsed and EventDiverted. Previously, in this race-condition scenario, SIP Server, while sending a re-INVITE message to the first leg and processing a 200 OK response received from the second leg, did not generate EventRouteUsed and EventDiverted. (SIP-27590)
If, after a switchover, DNS resolutions for SIP Proxy takes longer than for a Trunk DN resolution, SIP Server now uses the correct SIP Proxy address. Previously, SIP Server incorrectly used the address from the peer-proxy-contact option configured on a Trunk DN, which resulted in TCP exception and setting the Trunk DN to out of service. (SIP-27485)
When no-login-on-presence is set to true
, SIP Server now correctly processes an agent login and distributes EventAgentLogin followed by EventAgentNotReady. Previously, SIP Server sometimes rejected an agent login if a NOTIFY message with activities arrived from the Cisco UCM. (SIP-27349)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.26. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When, during an agent nailed-up connection, an inbound leg initiates a re-INVITE transaction, and then the dialog is terminated by a BYE message without acknowledging (ACK) the 200 OK (INVITE) transaction, SIP Server now correctly restores the nailed-up connection with the agent. Previously, in this scenario, SIP Server set an agent to Not Ready state and did not restore the nailed-up session with the agent. (SIP-27512)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.26. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
In SIP Cluster, when call routing to a reserved agent fails, and this call is routed with the agent reservation (using iscc-ar extensions) and established with another agent, who transferred the call to a Routing Point using a TSingleStepTransfer request, SIP Server now correctly sets the AttributeCallState to 1 (Transferred)
for this call in an EventQueued message. Previously, SIP Server incorrectly set the AttributeCallState to 0 (OK)
, which resulted in incorrect history reporting. (SIP-27493)
SIP Server is built with TSCP version 8.5.000.26, which corrects the following issue:
SIP Server now uses the correct name of the Configuration Server instance in alarm and log messages. Previously, SIP Server printed the default Configuration Server name confserv in alarm and log messages. (SIP-27443)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When an established call with recording (record=true
) is transferred to a Routing Point where a treatment is applied, and an agent sends a TSendDTMF request, SIP Server now correctly processes the request and generates an EventDTMFSent message. Previously, in this scenario, SIP Server dropped the call. (SIP-27467)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When a DN is configured with the dual-dialog-enabled option set to single-dialog-rtp-on-hold
and a consultation call is made from that DN to a Routing Point where a treatment is played, SIP Server now correctly handles the alternate operation between the main and consultation leg. Previously, SIP Server did not handle the call alternate operation correctly, which resulted in a dropped call. Limitation: SIP Server does not apply silence to a call during the hold procedure while the call is on a Routing Point where a treatment is played. (SIP-27439)
When a DN is configured with the dual-dialog-enabled option set to single-dialog-rtp-on-hold
and an agent places a call on hold and then initiates as a single-step transfer of the call, SIP Server now correctly processes the call transfer. Previously, SIP Server rejected that transfer request with an error message. (SIP-27420)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In a multisite transfer completion scenario using the SIP REFER method with the Replaces header, SIP Server now correctly resolves the SRV contact of the transfer destination. Previously, when SIP Server resolved an SRV record into a transport with a non-default SIP port, SIP Server incorrectly overwrote the SIP port with the default 5060 value in the request URI of the INVITE message, which prevented the two-step transfer completion. (SIP-27409)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new functionality:
This release includes the following corrections and modifications:
An inbound call no longer fails when it arrives through a trunk that is configured with the sip-ring-tone-mode option set to 1
and SIP Server processes the call with a preview interaction before routing the call to a DN, which responds first with an unreliable provisional response and then with a reliable response containing an SDP. Previously, in this scenario, the call failed. (SIP-27416)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
After an agent has established a call and Orchestration Server (ORS, or a T-Library client) has sent a RequestClearCall message, SIP Server now correctly reports that the ReleasingParty key has been set to 1 Local
in EventReleased on the agent DN. Previously, in this scenario, SIP Server reported that the ReleasingParty key had been set to 3 Unknown
when the releasing-party-report option was set to true
. (SIP-27395)
SIP Server clears internal DN ownership information when a corresponding DN is deleted from the Configuration Database. Previously, DN ownership information sometimes became stuck even after the DN was deleted and re-created in the Configuration Database.
To change the ownership type of a DN, delete the DN, wait for about 60 seconds, and create a new DN with a new contact option value. For example, if the contact option is set to a value of * (an asterisk), it is a SIP-based ownership type. (SIP-27508)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now processes the following consultation-call scenario correctly: an inbound call leg sends the first re-INVITE message when a hold procedure is not completed, and then it sends the second re-INVITE message when a transfer origination endpoint has not responded yet to an INVITE message for a new dialog. Previously, the consultation call failed in this scenario.
The sip-reinvite-action option, when set to after-hold
, now works correctly if the re-INVITE message is not empty (contains an SDP). (SIP-27446)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
For DNs configured behind a softswitch, SIP Server no longer sends an incorrect AttributeAddressType value in call-monitoring events, causing incorrect reporting of DN activities. (SIP-27428)
The SIP Server SP_NTLIBCLIENTS operational metric, representing the number of T-Library clients connected to the Smart Proxy module of SIP Server, is now updated correctly. Previously, the SP_NTLIBCLIENTS metric incorrectly remained set to 1
regardless of how many T-Library clients were connected to the Smart Proxy. (SIP-27423)
While processing a TCompleteConference request, SIP Server now correctly merges the main and consultation call legs in the following scenario:
Previously, because of a race condition, SIP Server did not complete the merging of the conference with the monitored call. (SIP-27405)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.23. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server operating in backup mode no longer terminates unexpectedly while processing stuck-call synchronization from the primary SIP Server. (SIP-27168, TS-11987)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In a Business Continuity deployment, in a multisite call monitoring scenario, where a supervisor on site 1 issues a TListenDisconnect request to suspend a monitored agent on site 2, SIP Server now correctly reports only the agent party state as ListenDisconnectedHeld. Previously, SIP Server incorrectly reported both agent and customer parties states as ListenDisconnectedHeld. (SIP-27316)
In the following scenario, Agent2 can now accept new calls:
Previously, after Step 8, SIP Server rejected all calls to Agent2 with the following message: another call already exists on this DN
. (SIP-26960)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When two clients are connected to the Smart Proxy module of SIP Server and registered to the same Trunk Group DN, the Smart Proxy now continues sending events for the Trunk Group DN to connected clients even if one of the clients disconnects. Previously, if a client that was registered to a Trunk Group DN client disconnected, SIP Server did not send related events to other clients registered to the same Trunk Group DN. (SIP-27191)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
While completing the Out Of Signaling Path transfer by the REFER method with the Replaces header, SIP Server now waits for a NOTIFY message with the final response event, even if the transfer destination terminates its dialog (according to RFC 5589). To work around this situation, set the sip-transfer-complete-message configuration option to 200
(the new supported value) on the Trunk DN representing the referred party.
There is also a new Application-level option sip-transfer-complete-timeout.
sip-transfer-complete-timeout
Setting: [TServer] section, Application level
Default Value: 0
(unlimited wait, for backward compatibility)
Valid Values: 0-34
Changes Take Effect: Immediately
Specifies how many seconds SIP Server waits for a NOTIFY message before considering the Out Of Signaling Path transfer as failed.
(SIP-27088)
This release includes the following corrections and modifications:
When sip-enable-sdp-codec-filter is set to true
and one of the m= lines in an SDP message body contains disabled media,
SIP Server sends the SDP without filtering the disabled media line out. Previously, SIP Server filtered the disabled media line out based on the option setting. (SIP-27294)
When the SIP_HEADERS extension is configured with a Diversion header and one or more additional headers, SIP Server now properly sends the Diversion header in a 302 Moved Temporarily response. Previously, SIP Server did not include the Diversion header in a 302 response. (SIP-27234)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while writing logs to a disk with slow performance. (SIP-26980)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
SIP Server supports the following new configuration option:
no-login-on-presence
Setting: [TServer] section, Application level
Default Value: false
Valid Values: false, true
Changes Take Effect: Immediately
When set to true
and, after receiving a SIP NOTIFY presence message, SIP Server does not log an agent into a desktop. A Ready/Not Ready state can be changed by a NOTIFY message. If using this option, Genesys recommends setting it before a SIP Server application is started.
(SIP-27162)
SIP Cluster mode: SIP Server supports Workspace Web Edition (WWE) 9.0 smart failover to another region. This feature preserves the agent state at repeated agent login sessions with the same credentials. When an agent desktop temporarily disconnects from a SIP Cluster node and then reconnects, it sends a TAgentLogin message for an already logged-in agent. This message could be sent to the same SIP Cluster node or to a node at another region. In both cases, the current agent login session is preserved with all parameters.
In multi-regional SIP Cluster, this feature relies on the Agent state restoration on DN ownership move feature—introduced in 8.1.103.45, SIP-24886— which must be enabled by setting the agent-state-auto-restore option to true
on the VOIP Service DN with service-type=sip-cluster-nodes
. The Agent state restoration on DN ownership move feature provides transition of the entire agent session to another node when the DN ownership is moved.
(SIP-26844)
This release includes the following corrections and modifications:
When SIP Server receives a ReasonCode value in AttributeReasons of TAgentNotReady from an agent desktop, SIP Server now reports this ReasonCode in AttributeReasons of EventAgentNotReady, when an AgentLogin session is restored on another SIP Cluster node because of the DN ownership move. Previously, SIP Server reported the ReasonCode in AttributeReasons before the DN ownership move and in AttributeExtensions after the DN ownership move. (SIP-27318)
SIP Server now preserves agent session parameters (the state, wrap-up-time, work-mode, reason) after repeated agent logging (TAgentLogin) for an already logged-in agent with the same credentials. Previously, if SIP Server received a TAgentLogin message when an agent was already logged in, SIP Server set the agent state and other agent session parameters to their default values. (SIP-27317)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.22. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When a pending transaction is in progress on a linked SIP dialog, SIP Server no longer rejects a re-INVITE message for a call that contains disabled media in the SDP. Previously, in this scenario, SIP Server sometimes rejected the re-INVITE message with a 503 response. (SIP-27250)
When SIP Server is in Cluster mode, it no longer allocates more memory when a client that is registered to a large number of DNs (8000+) disconnects. Previously, SIP Server allocated more memory while processing the large number of DNs from the disconnected client. (SIP-27177)
If SIP Server receives a 302 message when a new party is added to a single-step conference, it no longer sends an EventError message and continues connecting the redirected party. Previously, in this scenario, SIP Server sent an EventError. (SIP-27166)
For call supervision scenarios, when the new update-ctrl-party option is set to true
, SIP Server sets AttributeCtrlParty in EventCallDeleted to the party that has actually released the call. Previously, when an agent released a supervised call, SIP Server set AttributeCtrlParty to a customer DN, and when a customer released the call, SIP Server set AttributeCtrlParty to an agent DN.
update-ctrl-party
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
For call supervision scenarios, when set to true
, SIP Server sets AttributeCtrlParty in EventCallDeleted to the party that has released a call.
(SIP-26984)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
In standalone mode, SIP Server can now restrict SIP endpoint registration if its IP address is not included in a list of trusted IP addresses. When SIP Server receives a SIP REGISTER request from a SIP endpoint, it verifies the endpoint's IP address. You configure a list of trusted addresses using the sip-registrar-allowlist configuration option. If the REGISTER request is arrived from an untrusted IP address, SIP Server rejects the request with an error code defined by the sip-registrar-reject-code option.
sip-registrar-allowlist
Setting: [TServer] section, Application level
Default Value: An empty string
Valid Values: A string
Changes Take Effect: At the next registration request
This option contains a list of IP addresses, separated by a semicolon (;). An empty value means that this functionality is disabled.
Each entry in the list can be in one of the following forms:
For the FQDN and IP address entries, SIP Server makes an exact match of the entry to the address extracted from the REGISTER request. For a CIDR block, SIP Server takes n bits starting from the left of the address and matches them against n left bits of the entry. For example, to accept the range of 255 addresses from 192.0.2.0 to 129.0.2.255, the entry in the list must be as follows: 192.0.2.0/24
.
sip-registrar-allowlist-origin
Setting: [TServer] section, Application level
Default Value: via
Valid Values: via, contact
Changes Take Effect: At the next registration request
This option defines a REGISTER message header from which SIP Server takes an IP address to match against a list of IP addresses defined by the sip-registrar-allowlist option.
sip-registrar-reject-code
Setting: [TServer] section, Application level
Default Value: 403
Valid Values: A valid SIP error code in the range of 400-599
Changes Take Effect: At the next registration request
This option defines an error response that SIP Server sends if a SIP endpoint's IP address in its REGISTER request does not match the one defined in the trusted IP addresses list (in the sip-registrar-allowlist option).
(SIP-27103)
This release includes the following corrections and modifications:
When the Preview Interactions feature is enabled, SIP Server now correctly diverts calls according to the Call Divert Destination feature (a post-call survey). Previously, SIP Server ignored the after-call-divert-destination setting when a call was routed with a preview interaction. (SIP-27190)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
SIP Server now supports a new Application-level configuration option:
clearcall-sip-reject-code
Setting: [TServer] section, Application level
Default Value: 603
Valid Values: A valid SIP error response in the range of 400-669
Changes Take Effect: Immediately
Specifies the SIP code that SIP Server distributes while performing a ClearCall procedure for an inbound SIP call leg in the calling state.
(SIP-27152)
SIP Server's Smart Proxy module can now suppress distribution of EventAttachedDataChanged for its T-Library client (Stat Server), when Smart Proxy receives the SuppressUserData=1 KVP from that T-Library client in AttributeExtensions of RequestRegisterAddress messages.
In addition, the suppress-event-network-private-info configuration option is introduced for a VOIP Service DN with service-type=sip-cluster-nodes
to disable distribution of EventNetworkPrivateInfo messages by Smart Proxy to its T-Library clients.
suppress-event-network-private-info
Setting: [TServer] section, DN level
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Specifies whether EventNetworkPrivateInfo messages are distributed by Smart Proxy of the SIP Cluster Node to its T-Library clients. If set to true
, no EventNetworkPrivateInfo messages are distributed.
(SIP-27151)
This release includes the following corrections and modifications:
SIP Server no longer grows in memory while processing NOTIFY messages from GVP. (SIP-27150)
In a callback scenario, where the INFO message with a CPD response from the media server is delayed and received only after the call is cleared (because of a failed treatment followed by a failed parking of the caller), SIP Server now sends a 200 OK message and ignores that INFO message. Previously, SIP Server attempted to process the INFO message and terminated unexpectedly. (SIP-27100)
If the override-from-on-conf option is set to true
on a new party DN added to a single-step conference and an INVITE message is sent to the new party to create the conference, SIP Server now replaces the userpart of the From header with the DN or cpn
, instead of "msml=
" or "conf=
", even if the override-domain-from option is set. Previously, if both options were configured, SIP Server did not replace the userpart of the From header. (SIP-27087)
When options record-consult-call=true
and monitor-consult-call=true
, and an established call monitored by a supervisor is transferred to another agent, SIP Server now correctly disconnects the supervisor from the call when the transfer is competed. Previously, SIP Server incorrectly connected the supervisor to the caller after the transfer was completed. (SIP-26927)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following features or functionality:
This release includes the following corrections and modifications:
SIP Server no longer rejects new calls after a switchover when the switchover-on-trunks-oos option is set to true
. Previously, in a rare race-condition scenario, SIP Server sometimes rejected new calls after the switchover. (SIP-27021)
When the disable-media-before-greeting option is set to true
and a greeting fails to connect and play to a caller, SIP Server now correctly reconnects the caller and an agent with their respective recordings. Previously, after a greeting failure, SIP Server reconnected only the agent with the recording, but did not reconnect the caller with the recording. (SIP-26903)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server in Cluster mode now responds with a 404 Not Found message when a call is made to a Routing Point that was disabled in the configuration database. Previously, SIP Server erroneously distributed EventDNBackInService and EventUnregistered messages for the disabled Routing Point. (SIP-26986)
When the dual-dialog-enabled option is set to single-dialog-rtp-on-hold
, SIP Server no longer terminates a call upon receiving a refresh re-INVITE message from a party to which music on hold is played. (SIP-26963)
When the reject-call-in-call option is set to true
, SIP Server no longer rejects new calls to a logged-in agent with the "another call already exists on this DN
" error message when there is no active call for that agent. Previously, if SIP Server received an 18x provisioning response from a trunk for a call and then received an error response, that call became stuck at the agent DN, and SIP Server would reject any new calls. (SIP-26669)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
SIP Server in Cluster mode no longer incorrectly logs out an agent with an active call. Previously, because of the inactivity timeout set by the auto-logout-timeout option, SIP Server sent EventAgentLogout even if the agent had an active call. (SIP-26965)
When SIP Server receives a 486 Busy Here SIP response from a DN, it now correctly distributes EventError containing ErrorCode 231 and ErrorMessage DN is Busy
. Previously, SIP Server distributed EventError containing ErrorCode 100 and ErrorMessage Unknown Cause
for a busy DN. (SIP-26857)
When the keep-mute-after-conference and sip-enable-two-party-mute options are set to false
and RequestSetMuteOn is applied to a three-party conference, SIP Server now correctly generates an EventMuteOff message when one of the parties leaves the conference. Previously, SIP Server did not generate EventMuteOff. (SIP-26801)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server in Cluster mode now supports a new Application-level configuration option, cluster-acw-persistent-reasons. When it is set to true
, SIP Server includes AttributeReason with the ReasonCode key-value pair in EventAgentNotReady, which SIP Server distributes when the AfterCallWork timer expires. Previously, the ReasonCode parameters of active calls were not correctly synchronized after a SIP Server switchover.
cluster-acw-persistent-reasons
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true
, SIP Server includes AttributeReason with the ReasonCode key-value pair in EventAgentNotReady, which SIP Server distributes when the AfterCallWork timer expires. This is the same ReasonCode pair which was set in RequestAgentNotReady while a call was active. When set to false
(the default), this functionality is disabled.
(SIP-26928)
SIP Server now correctly retrieves a call from hold (the main or consultation call), when, during RequestAlternateCall processing, the active call (the main or consultation call) is released. Previously, SIP Server did not retrieve the call properly. (SIP-26883)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
t-library-stats-enabled
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: After SIP Server restart
When set to true
, SIP Server collects T-Library client statistics for SIP Server threads and embeds them in HTTP monitoring statistics. When set to false
(the default), this feature is disabled.
(SIP-26438)
SIP Server now supports CounterPath Bria v6.1.0, including audio only. (SIP-26770)
This release includes the following corrections and modifications:
When the sip-error-overflow option is set on a supervisor DN and a call recording is enabled, SIP Server now correctly invites a supervisor for call monitoring. Previously, starting with version 8.1.103.77, the call monitoring might fail in this scenario. (SIP-26877)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
SIP Server in Cluster mode can now optimize Smart Proxy logs (1792) to print T-Library events that are sent to or received from Smart Proxy clients in a summarized format. To enable this feature, set the new restricted smart-proxy-verbose-level option on the SIP Cluster Node DN (service-type=sip-cluster-nodes
).
smart-proxy-verbose-level
Setting: [TServer] section, DN level
Default Value: brief
Valid Values: brief, detailed
Changes Take Effect: Immediately
Enables Smart Proxy logs optimization. The option must be set on the SIP Cluster Node DN (service-type=sip-cluster-nodes
).
(SIP-26840)
This release includes the following corrections and modifications:
While operating in SIP Cluster mode, SIP Server now correctly distributes T-Events to its bulk registrant T-clients, such as Stat Server, via a Smart Proxy connection after a SIP Server switchover. Previously, SIP Server sometimes stopped distributing T-Events if its connection to Smart Proxy on another node was disconnected and restored while SIP Server was in the backup state. (SIP-26870)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.18. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When the sip-enable-sdp-codec-filter option is set to true
and SIP Server receives Content-Type multipart/mixed in an INVITE message for a call, SIP Server now correctly invites a Media Server to apply a treatment to the call. Previously, SIP Server did not send an INVITE message for a call treatment and generated an EventTreatmentNotApplied message. (SIP-26814)
If, in a multisite scenario, call routing from a local site to a destination site fails, SIP Server now reports an ISCC (multisite) transaction as terminated. Previously, SIP Server incorrectly reported this transaction as completed. (TS-11928, SIP-26751)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
When a THoldCall request fails before a TInitiateTransfer request is completed, SIP Server now properly completes the TInitiateTransfer call processing and successfully executes the next T-Library request from the queue. Previously, SIP Server did not properly process TInitiateTransfer and any following T-Library requests from the queue. (SIP-26499)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now responds with a 200 OK message containing the last known SDP in the following scenario:
SIP Server no longer terminates unexpectedly while receiving a TReleaseCall request from an agent during a recorder setup (an INFO message).
The issue occurred in the following configuration: sip-link-type=0
, record=true
, greeting-delay-events=true
, and a call routing with greetings. (SIP-26733)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
If SIP Server is connected to Configuration Server Proxy and, even when Configuration Server Proxy terminates, SIP Server no longer terminates unexpectedly, while accessing the missing configuration (CFG) objects. (SIP-26672)
When an agent initiates a consultation call and the main recording-enabled call is placed on hold and then failed because of the GVP failure, SIP Server now clears this scenario and completes the transfer successfully. Previously, SIP Server did not complete the transfer operation. (SIP-26426)
In a race-condition scenario, when RequestAlternateCall is in progress and a consultation call destination/origination releases the call, then, based on when the release happens, SIP Server retrieves the main call or generates an event error. Previously, in a similar scenario, SIP Server did not retrieve the main call when a consultation call party disconnected, and, in some scenarios, an agent was not able to retrieve the main call manually. (SIP-24992)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
While operating in SIP Cluster mode, SIP Server's Smart Proxy module now correctly distributes T-Events to its clients after a switchover. Previously, SIP Server's T-Controller and Smart Proxy modules were not correctly synchronized after a SIP Server switchover. (SIP-26700)
SIP Server now correctly routes a call that is transferred from a Routing Point back to the same number it arrived from. Previously, SIP Server incorrectly disconnected the call in this scenario. (SIP-26696)
If "Nailed-up Connection on Agent Login" is enabled and a nailed-up connection was terminated for any reason, SIP Server now correctly reconnects the nailed-up connection when an After Call Work (ACW) timer expires and an agent returns to the Ready state. Previously, SIP Server did not reconnect the nailed-up connection in this scenario. (SIP-26676)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
In a Business Continuity environment, SIP Server now places a DN located behind the media gateway back in service when a DR peer trunk returns in service. Previously, when a DR peer trunk returned in service, but the gateway trunk was still out of service, the DN remained out of service. (SIP-26647)
SIP Server no longer places a DN out of service before its registration expires. Previously, under rare conditions, SIP Server did not update the registrar timeout after receiving the new registration and used the expiration time of the first registration. (SIP-26602)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
SIP Server can now resolve a Fully Qualified Domain Name (FQDN) specified in the contact option of a DN using the asynchronous DNS resolution method and place the DN out of service if the FQDN is unresolvable. The feature applies to DNs of type Extension, ACD Position, and Voice Treatment Port. The DN-level enable-async-fqdn-resolve configuration option enables this feature.
enable-async-fqdn-resolve
Setting: [TServer] section, DN level
Default Value: false
Valid Values: true, false
Changes Take Effect: After restart
Specifies whether SIP Server resolves an FQDN address contact using the asynchronous DNS resolution method. If set to false
, SIP Server resolves the FQDN using the synchronous DNS resolution method. If set to true
, SIP Server resolves the FQDN of a DN using the asynchronous DNS resolution method. If the FQDN is unresolvable, SIP Server places the DN out of service.
In addition, when any outbound UDP/TCP connection is established, and if the address is the FQDN, SIP Server resolves it using the asynchronous DNS resolution method. If an asynchronous DNS resolution is unresolvable, SIP Server uses synchronous DNS resolution for the call. SIP Server continues applying the asynchronous DNS resolution method for the next call on that DN.
The enable-async-fqdn-resolve option applies only when the Application-level options sip-enable-gdns and enable-async-dns are set to true
.
When performing DNS resolution asynchronously using the DNS library service, SIP Server does the following based on the result:
Feature Limitations:
(SIP-26485)
This release includes the following corrections and modifications:
SIP Server no longer terminates a nailed-up connection and routes a call successfully when SIP Server simultaneously sends a re-INVITE message to an agent DN (with the nailed-up connection) and processes a TRouteCall request to a respective agent. Previously, in this race-condition scenario, SIP Server incorrectly terminated the nailed-up connection and responded with an error to a TRouteCall request. (SIP-26603)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates a nailed-up connection during an outbound call when SIP Server simultaneously sends a re-INVITE message to an agent DN (with the nailed-up connection) to start recording and receives a BYE message from a called party. Previously, in this race-condition scenario, SIP Server incorrectly terminated the nailed-up connection. (SIP-26510, SIP-26489)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
To extend functionality that was introduced for General Data Protection Regulation (GDPR) scenarios in releases 8.1.103.28 and 8.1.103.78, SIP Server now supports passing AttributeExtensions of a TRouteCall request into AttributeExtensions of an internal TMakePredictiveCall request when an agent makes a manual outbound call.
SIP Server passes AttributeExtensions received in TMakeCall into AttributeExtensions of an internal TMakePredictiveCall request if there are no AttributeExtensions in TRouteCall. In other words, AttributeExtensions of TRouteCall take priority over AttributeExtensions of TMakeCall. (SIP-26597)
This release includes the following corrections and modifications:
SIP Server now correctly adds the replaces tag in the Supported header of 200 OK responses to the INVITE messages containing replaces in the Supported header and to re-INVITE messages, if the main dialog contains replaces in the Supported header. This will ensure that SIP Server sends an INVITE request containing the Replaces header only when the replaces capability is supported.
When sending outgoing INVITE messages, SIP Server now adds the replaces tag in the Supported header based on the sip-enable-replaces DN-level option configuration.
sip-enable-replaces
Setting: [TServer] section, DN level
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
This option applies only to outbound INVITE messages and works as follows:
true
, SIP Server sends the replaces tag in the Supported header in INVITE messages.false
, SIP Server does not send the replaces tag in the Supported header in INVITE messages.(SIP-26265)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.16. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
While applying music-on-hold, SIP Server now properly handles the situation when a party rejects the music re-INVITE message with a 491 response and simultaneously sends its own hold INVITE containing the active SDP with music. Previously, while resending the re-INVITE message, SIP Server sometimes mistakenly sent a hold SDP. (SIP-26557)
SIP Server in Cluster mode now reads configuration about all nodes and then notifies the Smart Proxy and T-Controller threads, to ensure the configuration is read only once for each DN update. Previously, when the Smart Proxy did not connect to the T-Controller because of a failed configuration request, SIP Server sometimes did not update agent statuses, negatively impacting historical reporting of data. (SIP-26528)
When the monitor-party-on-hold option is set to false
and the sip-enable-call-info option is set to true
, and a supervisor intrudes into a consultation call with the agent scope, resulting in muting of external callers that are placed on hold, SIP Server now properly handles the mute/unmute operation. Previously, after the merge operation when a conference was established, SIP Server incorrectly muted the external caller, and others did not hear the muted caller in the conference call. (SIP-26479)
In a multisite deployment, SIP Server no longer distributes duplicate EventCallDataChanged messages while a consultation call is being established, which negatively impacted historical reporting of data. (TS-11896, SIP-26522)
SIP Server now correctly generates T-Library events for a routing destination in a consultation call if a 1pcc transfer is completed before the routing destination answers and the auto-answer is enabled on it (no 180 ringing response is received). Previously, in this scenario, SIP Server did not generate required T-Library events. (SIP-26514)
When a call is placed on hold briefly and then retrieved while the call is setting up for recording, SIP Server now correctly reconnects both parties to the recording. Previously, SIP Server did not reconnect the party that was on hold. (SIP-26259)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.15. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
SIP Server can now handle different HTTP error responses from SIP Feature Server for Dial Plan extended service (XS) requests in an enhanced way to address connection instabilities and provide a quality response to the origination side. See Enhanced handling of XS requests for details. The following new configuration options were added:
Note: This feature depends on support from a specific version of SIP Feature Server. Consult corresponding documentation for the availability of this new feature in that component.
(SIP-26145)
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while processing an INVITE message containing an unsupported JUR header in the NSS body. (SIP-26464)
When an agent initiates a transfer/conference request to an established consultation call and then sends a REFER with Replaces request to complete the transfer, SIP Server now correctly matches the Replaces dialog and successfully completes the two-step transfer. Previously, in this scenario, SIP Server incorrectly treated the REFER with Replaces requests as a single-step transfer and generated AttributeCause 11 in the EventReleased message for the consultation leg. (SIP-26463)
When a SIP Endpoint sends a consecutive REGISTER request over a TLS connection from a different port, SIP Server now sends the next call to a correct port. Previously, SIP Server sometimes sent the next call to the previous port, which resulted in calls being dropped. (SIP-26406)
The backup SIP Server now correctly restarts the cleanup-idle-tout timer during a call after receiving call synchronization events from the primary SIP Server. Previously, the backup SIP Server did not restart the timer and deleted the active call when the timer expired, and then recreated the active call incorrectly. (SIP-26384)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.15. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
To extend functionality that was introduced for General Data Protection Regulation (GDPR) scenarios in release 8.1.103.28, SIP Server now supports passing AttributeExtensions of a TMakeCall (with CPD) request into AttributeExtensions of an internal TMakePredictiveCall request and only those AttributeExtensions that are supported by the TMakePredictiveCall request.
The following DN-level option supports this feature:
make-call-cpd-extensions
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: BusinessCallType,SIP_HEADERS,CPNDigits,DisplayName,User-Agent
Valid Values: BusinessCallType,SIP_HEADERS,CPNDigits,DisplayName,User-Agent
Changes Take Effect: For the next call
Specifies a list of AttributeExtensions key-value pairs that SIP Server passes from a TMakeCall request in AttributeExtensions key-value pairs of an internal TMakePredictiveCall request. If no value is configured, the default value applies. If this option is configured, SIP Server passes only specified AttributeExtensions key-value pairs. When set to SIP_HEADERS
, SIP Server passes all key-value pairs that are present in the SIP_HEADERS
key.
(SIP-26284)
Enhanced disaster recovery solution for outbound calls: After receiving a negative response, SIP Server can now select an alternative trunk for outbound calls. In addition, SIP Server can now attempt to connect to a DN via an alternative softswitch (found in the DN configuration) if the first attempt to connect to a DN via a softswitch resulted in a negative response from that softswitch.
The following Application-level option supports this feature:
sip-error-codes-overflow
Setting: [TServer] section, SIP Server Application
Default Value: An empty string (or 503 error code)
Valid Values: A list of patterns for numeric error codes separated by a comma (,). Letter X in a pattern represents any digit. A single pattern must start with a digit and contain all 3 digits, and a pattern containing X should conclude the pattern's list, if present. Examples:
503
503,504
487,50X
487,5XX
5X3, XXX
are invalidWhen, on an initial INVITE message, SIP Server receives a negative response containing the error code that matches the option value setting, SIP Server attempts to find an alternative trunk or softswitch to initiate a call.
(SIP-26134)
There are no corrections or modifications in this release.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.15. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer encodes SIP headers when mapping T-Library request AttributeExtensions (for example, P-Asserted-Identity) to SIP headers. This fix applies only if the userdata-map-format option is set to sip-headers-encoded
. SIP Server encodes SIP headers when mapping the userdata to SIP headers. Previously, SIP Server encoded all custom SIP headers. (SIP-26412)
SIP Server now correctly associates a SIP dialog of an inbound call with a corresponding TLS transport. Previously, when SIP Server incorrectly associated a SIP dialog with the corresponding TLS transport, it sometimes resulted in lost outgoing in-dialog requests. (SIP-26396)
In a high-availability deployment, SIP Server no longer drops an active call after multiple switchovers. Previously, because of the network latency, the active call might not be synchronized properly on the new primary SIP Server, resulting in a call being dropped. (SIP-26313)
If a SIP Proxy hostname differs from the record located in the DNS (for example, the letter case does not match), SIP Server no longer selects a SIP Proxy in a round-robin fashion. Instead, for a device with transport=tls
, SIP Server now selects the SIP Proxy from which the REGISTER request was sent. (SIP-26272)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.10. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In a multisite configuration, if a connection to Configuration Server is lost during configuration load at SIP Server startup, SIP Server now correctly restores the connection to other SIP Servers. Previously, if the error occurred during configuration load, SIP Server sometimes ignored the configured connection to other SIP Servers, which could lead to problems with multisite routing. (SIP-25820)
When in SIP Cluster mode, SIP Server cancels the pending AfterCallWork (ACW) agent state after receiving the TAgentNotReady request containing WrapUpTime set to 0
in AttributeExtensions, and when the return-agent-on-call-to-post-acw-state option is set to true
, SIP Server now distributes a single EventAgentReady message. Previously, SIP Server distributed two EventAgentReady messages. (SIP-26377)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
In a Disaster Recovery (DR) peer forwarding scenario where dr-forward is set to no-agent
and the VOIP Service DN has charge-type set to 2
, when a conference is established on a Routing Point, and a call is routed to an agent by the REFER method, and that agent is not logged in to the local SIP Server, SIP Server now forwards the call to the peer site and the call is routed to an agent at that peer site. Previously, SIP Server did not forward the call to the peer site and generated a 603 Decline message. (SIP-26132)
In a multisite environment, where greeting, greeting-after-merge, and recording are enabled, and ISCC trunk optimization is disabled, SIP Server no longer terminates in some race-condition scenarios involving conferences and transfers. (SIP-25640)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
In a scenario where a remote agent located behind a PSTN trunk places a call on hold, SIP Server can now connect the on-hold party to a media service with a silent treatment to prevent disconnection of the call by the trunk. Previously, the connection was dropped because the inactivity timeout set by the trunk expired. This new SIP Server behavior affects only 3pcc Hold requests and cannot be applied to devices with dual dialog support.
To enable this feature, set the dual-dialog-enabled DN-level configuration option to a new value of single-dialog-rtp-on-hold
. In this case, SIP Server does not send an inactive SDP to the party during the hold operation. Instead, it connects the on-hold party to a media service with a silent treatment. A dual dialog is not allowed on that DN, only a single dialog is allowed, and it works the same as the option value of false
. (SIP-25864)
This release includes the following corrections and modifications:
When running in SIP Cluster mode, SIP Server now cancels the pending AfterCallWork (ACW) agent state after receiving the TAgentNotReady request containing WrapUpTime set to 0
in AttributeExtensions. Previously, SIP Server did not cancel the pending ACW agent state in this scenario. This fix applies only when the Application-level option return-agent-on-call-to-post-acw-state is set to true
. (SIP-26296)
When divert-on-ringing is set to false
and sip-treatments-continuous is set to true
, and an agent alternates the call while the consultation call is on a Routing Point treatment, SIP Server now correctly retrieves the main call and connects the required parties of the call. Previously, the consulted party was disconnected from the call when the main call was retrieved. (SIP-25993)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains the following new features and functionality:
This release includes the following corrections and modifications:
When SIP Server receives RequestAgentNotReady with AttributeExtensions containing WrapUpTime set to untimed
during a consultation call, SIP Server now correctly distributes EventAgentNotReady when releasing the call. A new Application-level configuration option, enhanced-pending-acw, must be set to true
to enable this SIP Server behavior. Previously, in this scenario, SIP Server did not distribute EventAgentNotReady.
enhanced-pending-acw
Setting: [TServer] section, SIP Server Application
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
When set to true
, SIP Server distributes EventAgentNotReady in the described above scenario.
(SIP-26252)
In a multisite call scenario in which a conference is initiated between two sites and, after the conference completion, an agent on the first site releases himself from the conference, and an agent on the second site mutes herself, SIP Server now handles the mute/unmute request properly and unmutes the agent on the second site. Previously, the call remained muted and SIP Server generated EventError for an unmute request. (SIP-26226)
When a call is routed to a DN from a Routing Point by means of a TSingleStepTransfer request, SIP Server now distributes EventRouteUsed with the correct AttributeReferenceID. Previously, SIP Server did not include AttributeReferenceID to EventRouteUsed. (SIP-26129)
When a call is routed to a Routing Point and the after-routing-timeout option expires, SIP Server no longer increments the CALLMANAGER_NROUTINGTIMEOUTS statistic. (SIP-25959)
SIP Server now includes X-Genesys-geo-location and X-Genesys-strict-location headers in each INVITE message to a conference leg to ensure that Resource Manager selects the right MCP for the conference. Previously, if recording was enabled for an agent, the INVITE message to the conference leg from the agent did not contain the geo-location headers. Similarly, if recording was enabled for a customer, the INVITE message to the conference leg from the customer did not contain the geo-location headers, which caused Resource Manager to select the wrong MCP. (SIP-25847)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
This is an update for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When processing a TSetMuteOff request for a call in an unmuted state, SIP Server now generates an EventError message and correctly processes a subsequent TSetMuteOn request. Previously, SIP Server generated an EventError message in response to the subsequent TSetMuteOn request. (SIP-26135)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When an agent's previous nailed-up connection is terminated and the agent then changes the status to Ready, and RequestAgentReady is generated containing the ReferenceID attribute, SIP Server establishes the nailed-up connection and now correctly includes the ReferenceID when generating an EventAgentReady message. Previously, SIP Server did not include the ReferenceID in the EventAgentReady message. (SIP-25996)
When SIP Server attempts to recover a greeting and a recording of the call, but the greeting recovery fails, SIP Server now attempts to recover the call recording as well. If SIP Server cannot recover the call recording, it now terminates the call recording session. Previously, if SIP Server did not recover the call recording, SIP Server incorrectly kept the call recording session active. (SIP-25833)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
For the ASM dialing mode, when an outbound call is merged using the bridging method, if an agent does a consultation call followed by a complete transfer, and call recording is enabled for both main and consultation calls, SIP Server now re-INVITEs a caller and the consulted agent properly when the transfer is completed. Previously, SIP Server attempted to reconnect a caller and a new recording leg bypassing SIP Proxy, which led to a call disconnection. (SIP-25997)
With the Call Pickup feature enabled, if a SIP endpoint has an established call and a new call is ringing on that endpoint, the ringing call can now be picked up by an available agent. Previously, when an available agent tried to pick up a ringing call on the endpoint with the established call, SIP Server responded with a 403 Forbidden error message. (SIP-25961)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server in SIP Cluster mode now supports passing the additional GVP parameters (which have Agent Assist supporting key-value pairs (KVPs) and Streaming KVPs) from AttributeExtensions of TRouteCall to MCP in the recording INFO messages, under existing recording metadata. The initial characters of recording GVP parameters must match the prefix specified in the new configuration option record-metadata-prefix. Those GVP parameters are added only if the following conditions are met:
record-metadata-prefix
Setting: [TServer] section, the SIP Server Application (takes priority) or the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: An empty string
Valid Values: Any valid string
Changes Take Effect: For the next call
Specifies the prefix that must match the initial characters of GVP parameters to be added from AttributeExtensions of TRouteCall in the INFO message sent to MCP. The matching KVPs are sent under recording metadata as additional GVP parameters with the prefix value stripped out. The setting at the Application level takes priority over the VOIP DN level setting. If this option is configured with an empty value at a VOIP DN, the existing recording metadata is sent without additional GVP parameters.
(SIP-25653)This release includes the following corrections and modifications:
SIP Server no longer terminates in a multisite conference/transfer completion scenario, where a remote site agent call type is changed to a Business call, in accordance with the options settings, and then the agent's state is changed from NotReady to Ready. (SIP-26056)
In a multisite call scenario in which a conference is initiated between two sites, if one of the agents mutes himself before the conference is established, SIP Server now handles the mute/unmute request properly and unmutes the agent when a conference is established. Previously, the call remained muted after a conference was established, and SIP Server generated EventError for a TSetMuteOff request. (SIP-26054)
If the monitor-party-on-hold option is set to false
and the monitor-consult-calls is set to true
, and an inbound call is connected to a supervised agent, and then two supervised calls (main and consult) merge into a regular two-way call after the call transfer completion, SIP Server now correctly connects two parties of the call. Previously, in this scenario, SIP Server did not properly connect the call to the transferee agent. (SIP-25985)
SIP Server now includes the trunk parameter in the request to SIP Feature Server when resolving a destination of an inbound call that arrived from an out-of-service trunk. Previously, SIP Server did not include the trunk parameter in the request to SIP Feature Server and, as a result, the inbound call destination replied with a 404 Not Found message. (SIP-25974)
SIP Server now clears SUBSCRIBE requests for Event: reg dialogs after moving to backup mode. Previously, SIP Server terminated unexpectedly while processing SUBSCRIBE dialogs when returning to primary mode after a double switchover. (SIP-25861)
To comply with RFC 3261, SIP Server now rejects an UPDATE request that is received for an established dialog with a 405 Method Not Allowed message. Additionally, SIP Server does not add the Allow header to an ACK request. Previously, SIP Server did not handle the unsupported UPDATE request for an established dialog correctly. (SIP-25831)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.5.000.06. TSCP is the shared software that all T-Servers and SIP Server use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of SIP Server.
There are no restrictions for this release. This release contains the following new information:
This release of SIP Server corrects the following Framework library issue:
SIP Server no longer terminates unexpectedly when a Routing Point DN is deleted from the configuration. Previously, when a Routing Point DN was configured with the list of Default DNs, which included that Routing Point, SIP Server terminated on deletion of that Routing Point DN. (SIP-25423)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
High-availability enhancement: The backup SIP Server can now check the status of its internal components during startup to ensure it provides the service when it is promoted to Primary. If any internal component (T-Controller, Smart Proxy, Interaction Proxy, or Operational Information thread) fails to complete initialization, SIP Server reports the SERVICE_UNAVAILABLE status to the Management Layer (LCA/SCS). This feature is covered in Verifying Initialization Status in Backup SIP Servers of the SIP Server High-Availability Deployment Guide.
In addition, if SIP Server fails to open a SIP port during startup, it reports the SERVICE_UNAVAILABLE status to LCA/SCS. Note that this approach does not work for the IP Address Takeover procedure or same host configuration.
To enable this functionality, configure the following options:
backup-init-check
Setting: [TServer] section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: After restart
When set to true
, SIP Server in Backup mode verifies that all internal components (T-Controller, Smart Proxy, Interaction Proxy, and Operational Information thread) are successfully initialized, and can provide the service when SIP Server switches to Primary mode. If some components fail to complete initialization, SIP Server reports the SERVICE_UNAVAILABLE status to LCA/SCS. The timeout for internal components to complete initialization is defined by the backup-init-check-timeout option.
backup-init-check-timeout
Setting: [TServer] section, Application level
Default Value: 60
Valid Values: 15-3600
Changes Take Effect: After restart
Restricted option. Specifies the timeout, in seconds, during which SIP Server verifies that all internal components are successfully initialized in a scenario described by the backup-init-check option.
backup-sip-port-check
Setting: [TServer] section, Application level
Default Value: true
Valid Values: true, false
Changes Take Effect: After restart
When set to true
, SIP Server in Backup mode attempts to open a SIP port. If the port opens successfully, no SIP messages are processed, and SIP Server closes the port immediately. If the SIP port does not open, SIP Server reports the SERVICE_UNAVAILABLE status to LCA/SCS. This functionality is enabled only when backup-init-check is set to true
. The functionality is disabled in the IP Address Takeover configuration, when the control-vip-scripts option is set to true
.
(SIP-25746)
This release includes the following corrections and modifications:
When running in SIP Cluster mode, and the tc-reconnect-timeout option is enabled, and T-Controller internal modules are disconnected between two nodes, SIP Server can now update DN ownership in a T-Controller on reconnection by sending a TPrivateService request containing AttributePrivateMsgID=8271. Previously, on reconnection of T-Controllers, the DN ownership was not updated correctly and, as a result, TRegisterAddress requests for DNs were rejected with EventError. (SIP-25979)
SIP Server no longer accepts a new connection when switching to backup mode. Previously, in a race-condition scenario, SIP Server accepted the new TCP connection and did not close it after switching to backup mode. (SIP-25655)
If a dialog is in early media mode and a PlayAnnouncementAndDigits treatment fails with an error response from Media Server, SIP Server now correctly sends an UPDATE request and connects a caller to Media Server to play the next treatment. Previously, SIP Server sometimes disconnected the call in this scenario. (SIP-25618)
When SIP stack initialization happens after an NIC failure, SIP Server's state is now correctly maintained and SCS in the Management Layer is notified about SIP Server's service restoration. Previously, SCS was not notified about SIP Server's service restoration. (SIP-24671)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server supports the following new configuration options:
sip-progress-response-code
Setting: [TServer] section, DN level
Default Value: 200
Valid Values: 100-699
Changes Take Effect: At the next call
Specifies the response code that SIP Server sends to an incoming re-INVITE or REFER message, which arrives when a dialog with Media Server is in progress. If set to 200
, SIP Server responds with a 200 OK message containing the latest SDP (backward compatible behavior). If set to a value in a range of 400-699
, SIP Server rejects a re-INVITE message with a respective error code.
sip-retry-after
Setting: [TServer] section, Application level
Default Value: 0
Valid Values: Any integer from 0-30
Changes Take Effect: At the next call
Specifies the value of the Retry-After header, in seconds, that SIP Server inserts in the error response to an incoming re-INVITE or REFER message, which is received while a dialog with Media Server is in progress. If set to 0
, SIP Server does not insert the Retry-After header.
(SIP-25901)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
As per GDPR requirement, on an outbound call made by a TMakeCall request through a Routing Point, SIP Server supports playing an opt-out prompt to an end user, so that the user can decide whether to opt out from recording and monitoring. This prompt can now be played in the customer's preferred language. The preferred language information can be provided in UserData attached to an engaged call by URS.
SIP Server passes the attached UserData from the engaging call to an outbound call. With this UserData available in the outbound call, SIP Server can pass the preferred language to GVP, while playing an opt-out IVR prompt for recording and monitoring to a customer. Previously, SIP Server did not pass the language information to GVP and a prompt was played in a default language.
To enable this functionality, at the Trunk Group DN level:
make-call-cpd-merged-userdata
Setting: [TServer] section, the Trunk Group DN
Default Value: No default value
Valid Values: A prefix, or a comma-separated list of prefixes that must match the initial characters of the key in the UserData key-value pair
Changes Take Effect: For next call
Specifies a prefix (or a list of prefixes) that must match the initial characters of the key in the UserData key-value pair. When the initial characters match, SIP Server passes the UserData key-value-pair from an engaging call to an outbound call. If this option is not specified, no data is mapped to an outbound call.
For example, if make-call-cpd-merged-userdata=test
and AttributeUserData contains 'test'='value1'
, 'testlocal'='value2',
and 'generaltest'='value3'
, only key-value pairs 'test'='value1'
and 'testlocal'='value2'
will be mapped. The 'generaltest'='value3'
will be ignored, because its initial characters do not match the prefix 'test'
.
Note: This option is enabled only on an outbound call made by a TMakeCall request with the GDPR feature enabled. To pass the filtered UserData in an outbound call to GVP, configure the userdata-map-filter option at the Trunk Group DN.
(SIP-25797)
This release includes the following corrections and modifications:
In a rare race-condition scenario, SIP Server no longer stops sending a SUBSCRIBE message to Resource Manager for a Trunk Group DN after a switchover. (SIP-25792)
In cases when the acw-persistent-reasons option is set to true
, a new configuration option, reset-acw-persistent-reasons, can be used to reset AttributeReason when any of the configured events are encountered.
reset-acw-persistent-reasons
Setting: [TServer] section, Application level
Default Value: No default value
Valid Values: agentlogout, agentready, all
Changes Take Effect: Immediately
If set to agentlogout
, SIP Server resets AttributeReason on receiving EventAgentLogout.
If set to agentready
, SIP Server resets AttributeReason on receiving EventAgentReady.
If set to all
, SIP Server resets AttributeReason on receiving EventAgentLogout or EventAgentReady.
(SIP-25730)
When running on a Linux machine with the installed openVPN client, SIP Server no longer terminates at startup because of a C library error. (SIP-25620)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server now supports enhanced high-availability resilience for network disruptions. With this feature, switchovers can be triggered when SIP Server detects MSML VOIP Service DNs and/or Trunk DNs that are out of service. After a switchover, a backup (new primary) SIP Server can place MSML VOIP Service DNs and/or Trunk DNs back in service and proceed with calls processing, minimizing or avoiding network outages.
switchover-on-msml-oos
Setting: [TServer] section, the SIP Server Application (in standalone mode) or the VOIP Service DN with service-type=sip-cluster-nodes
(in SIP Cluster mode)
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
Specifies the SIP Server action in case of losing connectivity to MSML VOIP Service DNs. When set to true
, in the case of strict matching only, VOIP Service DNs with the same or alternative geo-location are considered. After detecting that those DNs are out of service, SIP Server checks one more time that MSML VOIP Service DNs are unresponsive, before reporting the SERVICE_UNAVAILABLE status to LCA/SCS in order to trigger a switchover.
switchover-on-trunks-oos
Setting: [TServer] section, the SIP Server Application (in standalone mode) or the VOIP Service DN with service-type=sip-cluster-nodes
(in SIP Cluster mode)
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
Specifies the SIP Server action in case of losing connectivity to Trunk DNs. When set to true
, in the case of strict matching only, Trunk DNs with the same or alternative geo-location are considered. After detecting that those DNs are out of service, SIP Server checks one more time that Trunk DNs are unresponsive, before reporting the SERVICE_UNAVAILABLE status to LCA/SCS in order to trigger a switchover.
(SIP-25240)
This release includes the following corrections and modifications:
When a supervised call is sent to a Routing Point for an After Call Survey and then a TClearCall is issued, SIP Server now distributes an EventAbandoned message containing the ReleasingParty key set to a value of "1 Local
" in AttributeExtensions. Previously, SIP Server set the ReleasingParty key to a value of "3 Undefined
" in AttributeExtensions of EventAbandoned. (SIP-25794)
SIP Server now properly handles a refresh re-INVITE message for an Instant Messaging DN that is in hold state. Previously, SIP Server did not send a 200 OK response for that re-INVITE, which sometimes resulted in a dropped call when a dialog session expired. (SIP-25710)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When working in SIP Cluster mode and the reuse-sdp-on-reinvite option is set to true
, before executing the scheduled re-INVITE transaction, SIP Server now checks if a re-INVITE leg is already disconnected. That way, SIP Server does not send the INVITE message to the disconnected leg. Previously, in this race-condition scenario, SIP Server sent an INVITE message to a caller trunk instead of SIP Proxy. (SIP-25694)
While using the sips schema for secure SIP signaling, SIP Server now correctly forms the URI included in the SIP From header of the INVITE message sent to a switch. Previously, SIP Server incorrectly added the prefix of the port number to the URI in the From header. (SIP-25686)
When establishing a nailed-up connection with SIP authentication for DNs located behind the softswitch, SIP Server now responds with an INVITE message containing the Authorization header to the 401 Unauthorized response from a trunk. Previously, SIP Server did not respond with INVITE containing the Authorization header, preventing the connection from being established. (SIP-25604)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Per PCI compliance, SIP Server supports sending DTMF digits to an inbound leg of the call in SIP INFO requests. The call flow for this functionality is as follows:
gcti::reject
and AttributeExtensions with the following key-value pair: DtmfByInfo=<pin number>
.To configure this feature:
true
.true
on the inbound trunk DN that matches the external party of a call. That party must support the INFO request method (declaring it in the Allow header of a SIP message).dtmf
on an agent DN. If this DN option is configured, a SIP Endpoint sends DTMF packets in the format according to Endpoint's configuration. If this DN option is not configured, SIP Server involves MCP to generate RFC 2833 DTMF packets.(SIP-25661)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release improves music-on-hold handling in deployments where SIP Server is used with Secure Real-Time Transport Protocol (SRTP). Music continues playing if an agent places a call on hold. To enable this functionality, set the sip-reinvite-action configuration option to after-hold
on a Trunk DN where a re-INVITE message arrives.
sip-reinvite-action
Setting: [TServer] section, the Trunk DN, the VOIP Service DN with service-type=softswitch
Default Value: default
Valid Values: default, after-hold
Changes Take Effect: For the next call
Specifies how SIP Server processes a non-hold re-INVITE message from a party that is connected to the music-on-hold service while the process of placing a call on hold is not fully completed. When set to default
, SIP Server responds to the re-INVITE with 200 OK containing the latest known SDP. When set to after-hold
, SIP Server sends a 100 Trying message and waits for a hold procedure to be fully completed. After that the re-INVITE message is propagated to Media Server triggering a new SDP offer/answer exchange with Media Server.
(SIP-25480)
To troubleshoot T-Controllers (TC) communication issues, the following new statistics, based on periodic T-Library polls between T-Controllers, are now available:
You can configure the interval between poll cycles and TX_QUEUE_LEN/RX_QUEUE_LEN statistics using the tc-latency-poll-interval configuration option:
tc-latency-poll-interval
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: 10
Valid Values: Any positive integer
Changes Take Effect: For the next call
Specifies the time interval, in seconds, that each T-Controller polls its peers and updates *_QUEUE_LEN statistics. If set to 0
(zero), no polls are performed, all statistics are set to "disconnected value
" and neither statistic from this feature is updated.
(SIP-25476)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server in SIP Cluster mode now enforces authenticated access from SIP devices. To enable this functionality, set the sip-enable-strict-auth option to true
on the SIP Cluster Node VOIP Service DN. This option can also be set at the SIP Server Application or at DNs of type Extension.
sip-enable-strict-auth
Setting: [TServer] section
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next new call or REGISTER request
Enables SIP Server in SIP Cluster mode to mandate authorization of internal devices on REGISTER and INVITE requests. To register or establish communication, devices must not use empty passwords or passwords equal to the DN name. When this option is set to false
, an internal device can register or establish communication with SIP Cluster without any authorization.
You can define the sip-enable-strict-auth option at the following levels listed in order of priority:
sip-cluster-nodes
) (SIP-25101)
This release includes the following corrections and modifications:
When a DN located behind the softswitch is placed out of service, SIP Server now correctly sends SIP OPTIONS messages according to the oos-check option to recheck the DN state regardless of the SIP activity. Previously, if there was frequent SIP messaging for this DN, it remained out of service. (SIP-25651)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
0
for "Readiness for Switchover" in an overloaded condition (CPU usage is above the threshold).
When a backup SIP Server restarts and a primary SIP Server finishes sending missed call synchronization to the backup SIP Server, the backup SIP Server now periodically checks CPU utilization of SIP threads. If CPU usage of those SIP threads is below the threshold value (75%, by default) for a sufficient period of time (60 seconds, by default), then the "Readiness for Switchover" stat is reported as 1
. Previously, when a backup SIP Server restarted and, if its CPU usage grew to 100% (because of the increased T-Library request rate and call synchronization from the primary SIP Server), the backup SIP Server still reported the ''Readiness for Switchover'' stat as 1
. In this case, the switchover between SIP Servers might cause an error in processing of call synchronization and high CPU usage in the new primary (former backup) SIP Server.
This feature is enabled at a backup SIP Server when the Synchronization Complete notification is received from the primary SIP Server. The CPU check is done in the Network Monitoring thread of the backup SIP Server.
(SIP-25648)
none, false
: Disables the Preview Interaction protocol.tlib, true
: Enables preview interaction through T-Library messaging.chat
: Enables preview interaction through SIP Instant Messaging (IM). This release includes the following corrections and modifications:
SIP Server in SIP Cluster mode: When a DN is deleted from the configuration environment and then recreated again, SIP Server now correctly processes calls to that DN. Previously, after a DN was recreated, SIP Server considered such a DN as disabled and rejected requests to that DN. (SIP-25231)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server in SIP Cluster mode: A new Active Out-of-Service (OOS) check procedure for a VoIP service or Trunk DN ensures that this DN is placed out of service only when SIP Proxy processes the SIP traffic. Enable this procedure by setting the sipp-oos-recheck option to true
.
As soon as the SIP Proxy DN is placed from the in-service state to the OOS state, SIP Server initiates a switchover if the switchover-on-sipp-oos option is set to true
. This enables establishing a connection to SIP Proxy from another HA-peer application.
sipp-oos-recheck
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
If set to false
(the default), SIP Server behavior is not changed. If set to true
, SIP Server places a DN out of service only after rechecking that SIP Proxy is available, and does not place any other DN out of service if the SIP Proxy DN is already out of service.
switchover-on-sipp-oos
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
If set to true
, SIP Server attempts to initiate a switchover immediately after the SIP Proxy VoIP DN is placed out of service. The switchover is initiated by the SERVICE_UNAVAILABLE message sent to LCA/SCS. When SIP Server is switched to backup mode by SCS, SIP Server issues the SERVICE_AVAILABLE message to SCS.
(SIP-25130)
This release includes the following corrections and modifications:
When a subscription termination from a phone arrives to SIP Server after the new subscription is already established, SIP Server now correctly sends NOTIFY messages for a new subscription. Previously, SIP Server did not send NOTIFY messages for the SUBSCRIBE request sent for Hotelling events when a phone was restarted. (SIP-25535)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server running in SIP Cluster mode adds support for remote agents to use external numbers that are not provisioned in the Configuration Database. A new configuration option, replace-agent-phone, is introduced:
replace-agent-phone
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
SIP Server in SIP Cluster mode enables modification of dial digits that are required to reach an agent DN. This option setting affects all cluster internal DNs.
Feature limitations:(SIP-24839)
drop
. (SIP-25227)
This release includes the following corrections and modifications:
After a high-availability (HA) switchover, SIP Server now correctly synchronizes calls in the dialing state with a new primary SIP Server while processing a TMakeCall or TMakePredicitveCall request. Previously, the calls sometimes became stuck in the dialing state after a SIP Server switchover. (SIP-25255, SIP-25254)
SIP Server now correctly waits for a configured value of the predictive-call-timeout configuration option set to greater than 32 seconds during the ringing state of the outbound call and terminates the outbound call when the timeout expires. Previously, SIP Server did not apply the predictive-call-timeout option setting correctly. (SIP-25543)
When an inbound call arrives at an agent DN with a nailed-up connection and the INVITE message contains the X-Genesys header, if userdata-map-all-calls is set to true
, SIP Server now maps the SIP header into UserData correctly. Previously, SIP Server did not map call info for calls with nailed-up connections. (SIP-25459)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly in a scenario with the incorrect configuration in which the Application-level configuration option alternate-route-profile points to a Routing Point containing an empty Default DNs list, and the router-timeout expires at another Routing Point. (SIP-25546)
For DNs located behind the softswitch for 3pcc calls, SIP Server now correctly sends a DN name in the origin section of the request to SIP Feature Server that now includes agent-id in its response. Previously, SIP Server used the Request URI of the origination device instead of the DN name and SIP Feature Server did not include agent-id in the origin section of its response and, as a result, agent-id was missing in the INFO message sent by SIP Server to GVP for recording a 3pcc call initiated by a remote agent. (SIP-25534)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
na
on the Trunk DN to avoid playing music to an agent when a hold INVITE message arrives from the Trunk. (SIP-25174)
This release includes the following corrections and modifications:
In an active-active Resource Manager (RM) environment, if SIP Server receives an MSML error response from one RM to the INFO message for joining all the parties in a two-step conference, SIP Server now selects another RM to recover the conference. Previously, SIP Server did not establish connection with the second RM and failed to recover the conference. (SIP-25361)
When SIP Server tries to resolve any trunk with available gateways, and if an incoming trunk does not match its address with any MSML service, SIP Server no longer attempts to resolve those MSML services. Previously, SIP Server first resolved the MSML service and then tried to match the incoming trunk address with the MSML DN contact, which caused excessive logging. (SIP-25205)
When the session id of the origin header in the SDP becomes greater than a 32-bit integer value, SIP Server can still support that value and pass it correctly in the SDP message "o=" line. Previously, SIP Server increased the session id value from 9 to 20 digits, which became unacceptable for other SIP components. (SIP-25178)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
In SIP Cluster deployments, because of network issues, the connection between SIP Cluster nodes can be lost and promptly restored. T-Controller, the T-Library interface, maintains a subset of agent/DN states and interconnection to other T-Controllers in the cluster. A new configuration option, tc-reconnect-timeout, handles network outages gracefully in your environment.
tc-reconnect-timeout
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: 0
Valid Values: 0-60
Changes Take Effect: Immediately
Specifies the timeout, in seconds, during which T-Controller tries to reconnect to another SIP Cluster node when the connection between SIP Cluster nodes is lost. When the timeout expires and the connection is not restored, DNs owned by the disconnected T-Controller are declared out of service. The default value of 0
disables this functionality.
(SIP-24887)
In SIP Cluster deployments, where SIP-based DN ownership can be moved from one data center to another and, as a result, agent states are automatically changed to the logout state, SIP Server can now restore agent states immediately. A new configuration option, agent-state-auto-restore, handles agent states gracefully in your environment.
agent-state-auto-restore
Setting: [TServer] section, the VOIP Service DN with service-type=sip-cluster-nodes
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
Enables restoration of agent states when the DN ownership changes between SIP Cluster nodes. When set to false
, agents are logged out on the DN ownership change and must log in manually. When set to true
, agent states are restored on the new primary SIP Cluster node and agents are logged in automatically.
(SIP-24886)
This release includes the following corrections and modifications:
To achieve General Data Protection Regulation (GDPR) compliance, agents can now see the reason for the outbound call termination (before a customer answers the call) under the name "GSW_Outbound_Call_Result" on their desktop. If an outbound call is terminated with an error, such as NoAnswer, InvalidNumber, or Busy, SIP Server delivers an appropriate reason to the agent desktop for termination of the outbound call. Previously, SIP Server delivered the Unknown
reason in outbound call termination cases. (SIP-25393)
In multisite deployments where SIP Servers are connected via the SBC, SIP Server now sends a NOTIFY message containing the Event header in accordance with RFC 3857. Previously, SIP Server did not include the Event header, causing the SBC to reject the NOTIFY message with an error response. (SIP-25142)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now handles the MonitorMode and MonitorScope key values in TMonitorNextCall requests in a case-insensitive manner in both standalone and SIP Cluster modes. Previously, SIP Server handled those keys in a case-insensitive manner only in standalone mode. (SIP-25166)
SIP Server no longer grows in memory when a TMakePredictiveCall request (on a Trunk Group) is timed out or rejected with the 4XX error message. (SIP-25158)
When a subsequent monitored call with AssistMode=mute (after a first call with AssistMode=coach) is released, SIP Server no longer sets the supervisor to the NotReady state with WorkMode=ACW. Previously, when ACW was disabled but LegalGuard Timer was enabled for each call, the supervisor was incorrectly set to the NotReady state with WorkMode=ACW. (SIP-24995)
When the SDP that SIP Server receives contains two media lines—an "m=" line for active audio and an "m=" line for inactive video—SIP Server now correctly generates the hold SDP with the media line for inactive video. Previously, SIP Server incorrectly generated an empty "m=" line. (SIP-24961)
To comply with RFC 3262, SIP Server no longer adds the Contact header to the 200 OK message in response to the PRACK message. (SIP-24891)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In the Disable Media Before Greeting feature, SIP Server disables the SDP media for party A and party B before applying a greeting to both and enables the media after the greeting is completed. In this case, before enabling the media for party A, SIP Server now considers if party B has the hold media (intentional hold with the non-active media) and does not enable media for party A. Previously, though party B exchanged with the hold media, SIP Server still enabled the media for party A, which resulted in invalid exchange of SDPs between parties. (SIP-25070)
When SIP Server detects that a Trunk DN for an inbound call is out of service, to enable SIP Server to send a request using the transport associated with the dialog you must set the new configuration option, enable-retransmit-on-oos-transport, to true
at either an Application or DN level. Previously, without this option setting, SIP Server sent the timeout and terminated the call.
enable-retransmit-on-oos-transport
Setting: [TServer] section, Application or DN level
Default Value: false
(Application level); no default value (DN level)
Valid Values: true, false
Changes Take Effect: For the next call
When this option is set to true
, SIP Server continues retransmission of the SIP request using the transport associated with the dialog, even though the DN is detected as out of service. If set to false
, SIP Server does not retransmit the request and sends the timeout to the application layer when the DN is detected as out of service.
The DN-level setting takes precedence over the Application-level setting.
(SIP-24999)
When SIP Server detects that there is no active transport available for a destination and a SIP request is received through the transport associated with the dialog, SIP Server now places the transport in service and retransmits the SIP request using the same transaction. Previously, SIP Server created a new transaction for each retransmission request. (SIP-25011)
When a backup SIP Server disconnects from the primary, and fails to synchronize one or more dialogs of a main call, the corresponding main call and any consultation call for that main call will be deleted in the backup SIP Server. Previously, because of the sync failure, even after the main call and consultation call ended in the primary, the consultation call remained active in the backup SIP Server. (SIP-25007)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new configuration options:
sip-call-id-suffix
Section: [TServer] section, Application level
Default Value: sip-host
Valid Values: sip-host, sip-switch, sip-application
, any string, or empty
Changes Take Effect: After SIP Server restart
Defines the suffix that SIP Server inserts in the Call-ID header after the @ (at) character when SIP Server generates the INVITE message, as follows:
sip-host
(the default), SIP Server inserts the SIP listener IP address.sip-switch
, SIP Server inserts the name of the Switch object.sip-application
, SIP Server inserts the name of the SIP Server application.override-domain-ruri
Section: [TServer] section, DN level
Default Value: None
Valid Values: A non-empty string
Changes Take Effect: At the next call
Defines what SIP Server inserts in the host part of the Request URI.
sip-contact-user
Section: [TServer] section, DN level
Default Value: None
Valid Values: as-from
Changes Take Effect: At the next call
If this option is set to as-from
, SIP Server inserts into the Contact header the same user name found in the From header of the INVITE message.
sip-add-via
Section: [TServer] section, DN level
Default Value: None
Valid Values: peer-address
Changes Take Effect: At the next call
If this option is set to peer-address
, SIP Server adds the additional bottom-most Via header with the IP address of the peer SIP endpoint.
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If, while sending a refresh INVITE message, a TCP connection is lost between SIP Server and a call party, SIP Server now retransmits the INVITE through the new TCP connection. Previously, SIP Server did not retransmit the INVITE, which blocked further session refresh INVITE messages being sent to that party. (SIP-24976)
When trying to establish a third-party recorder connection for an outbound call, SIP Server now ignores unreliable 18x response messages, if the new option, sip-disable-unreliable-sdp, is set to true
on the Trunk DN for outbound calls.
sip-disable-unreliable-sdp
Setting: [TServer] section, DN level
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
When set to true
, SIP Server does not process an unreliable 18x response message containing the SDP received at the DN where this option is configured.
(SIP-24821)
Supported Operating Systems
New in This Release
Corrections and Modifications
Note: This version was first released as an Update on 02/22/19.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
While using the sips schema for secure SIP signaling, SIP Server now correctly forms the URI in the SIP From header by adding the TLS port. This includes predictive outgoing calls. Previously, SIP Server incorrectly included a non-TLS port in the From URI. (SIP-24962, SIP-24911)
In a multisite call, when the sip-enable-call-info option is set to true
and an agent does a single-step conference to a Routing Point followed by a TListenDisconnect request, SIP Server now disconnects the agent from the conference until the agent invokes a TListenReconnect request. Previously, the agent was not disconnected from the conference properly. (SIP-24936)
SIP Server now properly propagates mapped key-value pairs with empty values from UserData to SIP headers of generated INVITE messages. Previously, key-value pairs with empty values were filtered out and not propagated as mapped. (SIP-25115)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.13. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
0
to 1
. Now when the enable-async-dns option is not set in the common section of SIP Server applications, SIP Server processes DNS host name resolution asynchronously. (SIP-24469)
When running in SIP Cluster mode, SIP Server now communicates with the SIP Feature Server dial plan through a new Extended Services component (that is run in its own thread) instead of the Call Manager thread. The Extended Services component takes the Dial Plan request from Call Manager and returns the Dial Plan results after processing in SIP Feature Server. Communication with SIP Feature Server in the dedicated thread unloads Call Manager and decreases locking and waiting time in Session Controller. The External Services component produces its own log with suffix 2048.
When HTTP Monitoring is enabled, SIP Server provides monitoring information for the Extended Services component in the Extended Services section, as follows:
See HTTP Monitoring Interface for details. (SIP-24726)
This release includes the following corrections and modifications:
When SIP Server receives a TApplyTreatment request with the TEXT parameter pointing to an absolute file path, the path string must be appended with the parameter is-absolute-path with a value of true
. This way SIP Server does not remove the file://
string when passing the file path to MCP, and MCP is able to access the absolute file path. Previously, SIP Server removed the file://
string for both relative and absolute paths, and MCP was not able to access the absolute file path. (Example of the absolute file path: file://c:\MusicRecourses\GoodBuyGreeting.wav;is-absolute-path=true
.) (SIP-24814)
If the SIP Server HA pair fails to synchronize registration of a DN nailed-up connection during network instabilities, agents can now establish a nailed-up connection by logging out and logging in again, once the network becomes stable. Previously, if the ADDP was disabled between primary and backup SIP Server instances, agents were not able to establish a nailed-up connection after re-logging in. (SIP-24763)
In a non-MSML outbound deployment, when TMakePredictiveCall is requested to a trunk that is configured with the replace-prefix option and is not configured with the cpd-capability option, SIP Server properly creates the Request-URI of the INVITE message sent to the trunk. Previously, this Request-URI was formed incorrectly. (SIP-24753)
SIP Server now correctly adds the replaces tag in the Supported header of INVITE, re-INVITE, and 200 OK responses. Previously, SIP Server supported replaces by default, but did not explicitly add the replaces tag in the Supported header. That might cause to some SBCs to filter the Replaces header out, assuming that SIP Server does not support it. (SIP-24676)
When switch partitioning is enabled in SIP Server (the dn-scope option is set to a value other than undefined
), SIP Server no longer terminates while generating EventNetworkReached in the scenario where Agent1 conferences a call to Agent2, and Agent2 transfers the call to an external trunk, and then Agent1 releases from the call before the trunk sends confirmation. This fixed issue applied to the single-step conference and single-step transfer scenarios. (SIP-24787)
SIP Server no longer terminates a SIP dialog which provides music when an agent invokes a TListenDisconnect request while he or she is in a conference with a Routing Point. Previously, starting with version 8.1.102.77, SIP Server incorrectly terminated that SIP dialog and corresponding music treatment. (SIP-24774)
When running in SIP Cluster mode and when the http-port option is enabled, SIP Server no longer prints inaccurate messages about the SIP Server HA status in its log. (SIP-23116)
When an agent places a call on hold, but Media Server (to play music on hold) is not responding, then after the agent cancels the hold operation, SIP Server now sends a 487 Request Terminated message, re-invites both call legs, and updates the SDPs. Previously, in this scenario, call parties did not hear each other because of the incorrect SDP after canceling the hold operation. (SIP-24187)
When running in SIP Cluster mode, SIP Server now correctly processes the untimed
value for the WrapUp time that is set by UserData. Previously, starting with version 8.1.103.20, the agent had not been set to NotReady state after the call release in accordance with the configuration. (SIP-24789)
In a multisite environment, SIP Server now cancels an active ISCC transaction after receiving an error for the call routing attempt. Previously, the transaction was not canceled as expected, which caused external Routing Points to be blocked. (SIP-24784)
As of January 30, 2019, this release is no longer available. A critical issue was discovered. If you downloaded the software, do not install it. The identified issue will be addressed in the next release.
As of January 30, 2019, this release is no longer available. A critical issue was discovered. If you downloaded the software, do not install it. The identified issue will be addressed in the next release.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When a TClearCall request is invoked while a call is on a Routing Point during a post-call survey, SIP Server now distributes EventAbandoned with the ReleasingParty key in AttributeExtensions set to a value of 1 Local
. Previously, SIP Server distributed EventAbandoned with the ReleasingParty key set to a value of 3 Undefined
. (SIP-24733)
In the IP Address Takeover HA configuration, SIP Server runs a Virtual IP address script in accordance with the vip-state-change-timeout option configuration. Previously, if there were network problems and multiple switchovers occurred, SIP Server was not able to run the Virtual IP address scripts successfully. (SIP-24650)
In a SIP Cluster deployment, when an outbound call is done via a Routing Point, SIP Server now correctly adds the X-Genesys-Route header in the ACK message. (SIP-24648)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
true
in the Person object associated with the agent.true
in the VoIP Service DN containing the service-type option set to sip-cluster-nodes
.Note: In standalone mode, Agent Login objects are used to configure agent-based recording (record=true
). In SIP Cluster deployments, Agent Login objects are not used.
record
Setting: Annex tab, TServer section
Default Value: false
Valid Values: true, false
Changes Take Effect: When the next call is established on the DN
When set to true
and when a call is established on the DN on which this person has logged in, recording starts. SIP Feature Server reads this option and adds the option setting to the XS Dialplan Response. SIP Server reads the Dialplan Response and processes the call recording.
request-person-options
Setting: Annex tab, TServer section
Default Value: true
Valid Values: true, false
Changes Take Effect: When the next call is established
When set to true
, SIP Server adds the "request-person-options" tag to the XS Dialplan Requests that are sent to SIP Feature Server.
(SIP-23523)
This release includes the following corrections and modifications:
After multiple switchovers, SIP Server now correctly sends CTILINK messages. (SIP-24700)
An inbound call is now connected to an agent, even if SIP Server receives the refresh UPDATE requests in early dialog state. Previously, SIP Server did not connect the call to an agent. (SIP-24639)
SIP Server now correctly retransmits a 200 OK message for transport=tcp when the sip-rel-200-retransmit option is set to true
. (SIP-24740)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When establishing a nailed-up connection with an agent (connect-nailedup-on-login = gcti::park
), and if an MSML VOIP Service DN contains the contact-list option with a list of multiple IP addresses containing the transport=tcp parameters, SIP Server now correctly includes the transport=tcp parameter in the Contact header when sending an INVITE message to the agent. Previously, SIP Server did not include the transport=tcp parameter in the Contact header, defaulting to the UDP transport. (SIP-24590)
When a REFER with the Replaces header request is issued to perform a blind transfer to a Routing Point and a treatment is played, the backup SIP Server deletes the old media service and its associated connection when the call ends. This ensures that the DN is unreserved when a call is terminated, and when a switchover occurs, SIP Server processes the call on this agent DN. Previously, after a switchover, SIP Server did not process the call on the agent DN with the reject-call-incall option set to true
, because it remained in the reserved state. (SIP-24392)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes no corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
Hoteling subscriptions are now synchronized between SIP Servers in the HA pair, and SIP Server now increments a CSeq number for a specified method after each switchover. Previously, hoteling subscriptions were not synchronized, and when a double switchover occurred, the new primary SIP Server sent a hotelling NOTIFY message with the CSeq number less than the previous one, which was rejected by a Polycom phone. (SIP-24601)
Hoteling subscriptions are now synchronized between SIP Servers in the HA pair, and subscription dialogs are now correctly associated after each switchover. Previously, hoteling subscriptions were not synchronized and subsequent requests were rejected by SIP Server. (SIP-24544)
When blind-transfer-enabled is set to true
, sip-ring-tone is set to 1
, and recording is enabled on all 3 DNs, SIP Server no longer drops a call when a consulted party answers the blind-transferred call. (SIP-24339)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If call recording fails during call redirection to an after-call survey, SIP Server no longer generates an alarm and increments the NCALLRECORDINGFAILED statistic. (SIP-24485)
In a multisite deployment, when a remote observer connects to an observing conference and the observer's device sends several re-INVITE messages, SIP Server no longer disconnects the observer from the call. (SIP-24484)
In a multisite deployment, when an agent and a customer are located at one site, and if an observer that is located at another site places the call on hold, the agent and the customer now correctly hear silence. Ensure that the sip-enable-call-info option is set to true
. Previously, the agent and the customer heard music-on-hold instead. (SIP-24465)
In SIP Cluster mode, after a BusinessCall that a supervisor monitored is released, SIP Server now correctly applies the Application-level option call-monitor-acw and the AfterCallWork (ACW) state to the supervisor. Previously, SIP Server placed the supervisor to the ACW state even when the call-monitor-acw option was set to false
. (SIP-24450)
SIP Server no longer opens inter-thread TCP ports to external clients. Previously, SIP Server sometimes terminated when a network scanning tool probed internal SIP Server ports. (SIP-24421)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In a multisite deployment, after a call is routed to an agent on the second site and when the agent does a single-step transfer using the pullback transaction type while a re-INVITE request is in progress, SIP Server now performs the transfer operation after completion of the re-INVITE operation. Previously, in this race-condition scenario, the call was disconnected when the next TRouteCall was processed. (SIP-24481)
When running in SIP Cluster mode, if SIP Server receives a TRegisterAddress request for an Extension DN that is disabled in the configuration database, SIP Server now correctly responds with EventRegistered (status=OOS). Previously, SIP Server responded with EventError "DN is not configured in CME"
. (SIP-24166)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When a DN is configured with the force-register option to require registration, SIP Server now correctly processes a 407 Proxy Authentication Required
message that it receives in response to a REGISTER request. Previously, SIP Server ignored a 407 response and did not generate a new REGISTER request with authentication information. (SIP-24268)
SIP Server supports a new Application-level configuration option for the Agent Login and State Update on SIP Phones feature:
agent-allow-empty-password
Setting: [TServer] section
Default Value: true
Valid Values: true, false
Change Take Effect: For the next agent login
When set to true
, SIP Server allows an agent to log in from a SIP phone without the password. When set to false
, SIP Server rejects agent logging from a SIP phone without the password.
(SIP-21805)
When running in SIP Cluster mode, if the external-contact option is set to an SRV record, SIP Server now correctly resolves it and adds the correct Route header for outbound SIP messages. (SIP-24479)
When running in SIP Cluster mode, SIP Server no longer consumes excessive memory while synchronizing its messages in this rare scenario: SCS loses its connection to a primary LCA, SCS instructs the backup SIP Server to switch its role to a primary, and, at the same time, the primary SIP Server remains running in primary mode. (SIP-24016)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.12. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a single-step conference scenario from an agent to a recorder (gcti::record
), when the call is released by the agent, SIP Server now correctly includes AttributeCtrlParty in EventCallDeleted. (SIP-24249)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In multisite deployments, when a consultation call from Site 1 arrives as a non-business call at Site 2 (the bsns-call-dev-types option was set to -xrp
), and after a consultation agent at Site 1 completes the transfer of the consultation call, SIP Server now correctly changes the call type to business
if the main call at Site 1 was a business call. Previously, in such deployments, SIP Server did not set the call type to business
. (SIP-24189)
In multisite deployments, when reject-call-incall is set to true
, SIP Server now correctly handles a postponed scenario for a REFER request that it receives while another call-related operation is in progress. Previously, in this race-condition scenario, SIP Server did not clean up a transaction because of the postponed call operation, resulting in incorrectly rejected calls. (SIP-24008)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In deployments with SIP Proxy, after receiving a TNetworkPrivateService request containing a reboot parameter from SIP Feature Server for a registered device, SIP Server now correctly sends NOTIFY to that device to inform the device about its reboot. Previously, SIP Server did not send NOTIFY to the device. (SIP-24304)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support Genesys Cloud implementations only.
In this release, SIP Server, operating in the Cluster mode, modifies handling of pending AfterCallWork cancellation, controlled by a new Application-level option return-agent-on-call-to-post-acw-state.An agent enters pending AfterCallWork by issuing a TAgentNotReady request with AfterCallWork work mode while handling a call classified as business.
Pending work mode cancellation is executed by a TAgentNotReady request with AfterCallWork work mode and WrapUpTime extension with a value of 0
.
return-agent-on-call-to-post-acw-state
Setting: [TServer] section
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
This option modifies pending AfterCallWork cancellation in the following way:
true
, SIP Server restores an agent state that the agent was in prior to the pending AfterCallWork by distributing an unsolicited event corresponding to the restored agent state.false
, SIP Server does not return an agent to the previous state that the agent was in prior to the pending AfterCallWork.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a race-condition scenario when the ACK message with SDP Answer is delayed during SDP negotiation between two parties, SIP Server now correctly connects a call. Previously, if the called party sent a re-INVITE without the SDP and did not send the ACK message for the 200 OK with the last known SDP soon enough, the call became stuck. (SIP-24239)
When a REFER request is rejected with a specific error code, SIP Server now forwards that error code in the body of the NOTIFY message. Previously, SIP Server incorrectly sent a default error code "500 Server Internal error"
in NOTIFY, instead of the received error code. (SIP-24165)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When sip-signaling-chat is set to none
, and an inbound call with AtrributeExtentions chat set to true
is made to a Routing Point, and after receiving the TRouteCall request containing RouteTypeReject, SIP Server now correctly releases the call. Previously, SIP Server rejected TRouteCall with the EventError message. (SIP-24213)
When disable-media-before-greeting is set to true
, and a call containing invalid greeting extensions in AttributeExtensions of TRouteCall is routed to an agent, SIP Server now correctly connects an agent with a caller. Previously, SIP Server placed the agent on hold. (SIP-24191)
When an inbound call is routed to an agent and then returned to a Routing Point according to the no-answer-overflow option that is set to recall
, and a treatment is applied, SIP Server now applies the media stream correctly. Previously, if the SDP was sent in response to the first INVITE in an unreliable 180 Ringing message, SIP Server did not apply the media to a treatment correctly. (SIP-24132)
SIP Server now correctly sends EventAgentLogin and EventAgentReady messages when an endpoint with the enable-agentlogin-presence option starts in the out-of-service state and registers with SIP Server using the SIP REGISTER request. (SIP-24062)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a multi-site deployment, in a remote supervision scenario with a scope agent
, a caller from Site 1 is being transferred to the SIP Server at the remote site (Site 2) and a supervisor from Site 1 starts monitoring Agent 2 on Site 2. Previously, after the TCompleteTransfer request, the caller was added through the INVITE with Replaces message to the SIP Server at Site 2, and the supervisor's desktop was not updated with the caller number. Now, call participants are correctly updated on the supervisor's desktop at Site 1. (SIP-24002)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.10. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When enable-strict-location-match is set to true
and no-answer-timeout expires, SIP Server now correctly connects the caller to the agent's voicemail (no-answer-overflow). Previously, in this scenario, SIP Server did not connect the caller to the voicemail. (SIP-23936)
When SIP Server goes out of the signaling path, the custom headers from a REFER message are now filtered and sent in the outgoing REFER according to the sip-pass-refer-headers option setting. Previously, in this scenario, SIP Server did not filter custom headers from the REFER. (SIP-23190)
SIP Server now sends EventPartyChanged and EventUserEvent in the correct order when the epp-tout option is set to a non-zero value. Previously, SIP Server sent EventUserEvent with the new ConnID value before sending EventPartyChanged that contained the previous and actual ConnID. (SIP-23136)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in this release of SIP Server.
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly starts recording of a predictive outbound call when Agent A transfers the call to a Routing Point and the call is routed to Agent B. Agent DNs are configured with record=true
and the ringback. Previously, SIP Server incorrectly started recording during the ringback and did not reconnect the recorder to the call, which resulted in a dropped outbound call. (SIP-23791)
When an external Trunk does not respond to a TMakePredictiveCall request and SIP Server attempts to find an alternative Trunk without success, and the call is released because of the timeout, SIP Server now generates EventReleased containing CallState 7 (CallStateNoAnswer). Previously, in this scenario, SIP Server generated EventReleased containing CallState 0 (CallStateOk). (SIP-23513)
Note: This issue was initially fixed in release 8.1.103.00, but now extends to the alternative trunk scenario.
When a TRouteCall request contains customer-greeting and agent-greeting key-value pairs in AttributeExtensions, SIP Server now correctly applies greetings to a call in a scenario where divert-on-ringing is set to false
and the call is diverted to another agent DN because after-routing-timeout expires. Previously, in this scenario, SIP Server did not apply greetings. (SIP-23347)
Supported Operating Systems
New in This Release
Corrections and Modifications
Note: This version was first released as an Update on 04/17/18.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When the Masking Sensitive Data in SIP Messages feature is enabled, SIP Server no longer prints received DTMF symbols in the SIP Server log files. Previously, if the log output was set to the Debug level, some DTMF data was printed in the Debug-level messages. (SIP-23669)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When an inbound call is routed to an agent with the nailed-up connection and a preview interaction is enabled, SIP Server now extracts the AttributeExtensions from the TRouteCall request and apply them accordingly to the call after the preview interaction is finished. Previously, SIP Server did not extract the extensions from the TRouteCall request, which resulted in the wrong music-on-hold file being played. (SIP-23712)
When an agent is in a conference call for a call supervision and a consultation call is established at the remote site, and the record-consult-call option is set to true
, if the agent completes a transfer, SIP Server no longer sends a SIP INFO message containing the <modifyconference> element to the consultation call recording leg. Previously, SIP Server incorrectly sent the malformed MSML message that was rejected by MCP, which caused the transfer operation to fail. (SIP-23674)
Recording of the GSW_CONNECT_TIME value always occurs after the corresponding 200 OK is received by SIP Server. Now, SIP Server correctly shows the value of GSW_CONNECT_TIME. Previously, the GSW_CONNECT_TIME value was sometimes recorded 1 second earlier than the logged timestamp of the triggering 200 OK message. (SIP-23653)
When a nailed-up connection is configured with the Preview Interaction and reuse-sdp-on-reinvite is set to true
, and an agent with the nailed-up connection is connected to an inbound caller through aTRouteCall request for the second time, SIP Server now reports AttributeOtherDN as the inbound caller number in both EventRinging and Event Established messages. Previously, SIP Server incorrectly reported AttributeOtherDN as the Routing Point number in EventEstablished. (SIP-23460)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server supports a new Application-level configuration option for the integration with the RedSky emergency call provider.
sip-elin-timeout
Section: TServer
Default Value: 1200
Valid Value: 0
–3600
Changes Take Effect: For the next call
Specifies the time interval, in seconds, for SIP Server to keep in memory the association between a 911 caller and the Emergency Location Identification Number (ELIN) assigned to the caller. If a call arrives at that ELIN before the timeout expires, the call is sent to the associated 911 caller DN. If within this time interval there are several emergency calls with the same ELIN, SIP Server directs the callback to the latest caller.
(SIP-23610)
This release includes the following corrections and modifications:
SIP Server now correctly handles error responses from SIP Feature Server that are sent in fragmented messages. Previously, in this scenario, SIP Server sometimes terminated unexpectedly. (SIP-23636)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support Genesys Cloud implementations only. (SIP-23550)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When an inbound call is established with an agent and a consultation call is going through an outbound Trunk with required authentication, SIP Server now correctly sends an INVITE with authorization parameters and without the SDP. In addition, after the CompleteTransfer the agent leaves the call, both external parties have a proper RTP flow. Previously, in this scenario, SIP Server sent the old SDP content that resulted in a one-way audio call. (SIP-23553)
If the DN-level option use-register-for-service-state is set to true
and the registration expires, SIP Server now correctly places that DN in out of service. Previously, SIP Server placed the DN back in service when the recovery-timeout option expired, even if the DN contact was empty because of the expired registration. (SIP-23529)
When an external Trunk responds to a TMakePredictiveCall request with only a 100 Trying message (without 180 Ringing) and the call is released because of the timeout, SIP Server now generates EventReleased containing CallState 7 (CallStateNoAnswer). Previously, in this scenario, SIP Server generated EventReleased containing CallState 0 (CallStateOk). (SIP-23513)
Note: A new alternative trunk scenario was added in release 8.1.103.07. Use release 8.1.103.07 for your environment.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In a multi-thread SIP Server deployment configured with drop-nailedup-on-logout=true
, a nailed-up connection to an agent using a non-provisioned external phone number is now correctly terminated by SIP Server when the agent logs out. Previously, in this scenario, SIP Server did not terminate that nailed-up connection. (SIP-23388)
An agent is now able to log in from the Polycom phone using a new Agent Login ID after Configuration Server was restarted. Previously, in this scenario, SIP Server did not process the SUBSCRIBE message with the x-broadworks-hoteling
event from the Polycom phone properly. (SIP-23386)
In scenarios where SIP Server receives EventNetworkPrivateInfo from the other site to start recording, SIP Server no longer starts recording on a Routing Point where a call is still queued. Previously, SIP Server incorrectly started recording of the call that was queued on a Routing Point. (SIP-23339)
If a PlayAnnouncementAndCollectDigits treatment is stopped by GVP with an error response to the PlayAnnouncement dialog, SIP Server now correctly terminates both the associated treatment dialogs. Previously, SIP Server did not terminate the CollectDigits dialog when a PlayAnnouncement dialog was terminated by GVP, which resulted in collection of duplicated digits. (SIP-23325)
SIP Server now correctly connects an observer to a call using a TSingleStepConference request with AssistMode=coach. Previously, if the destination DN in the TSingleStepConference request from the observer was changed to a new DN by applying the dial-plan-rule option, SIP Server rejected further TSingleStepConference requests to the same DN with EventError. (SIP-23262)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while trying to reconnect to SIP Feature Server when one of the Feature Server nodes is down. (SIP-23463)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In the scenario where a conditional dial-plan parameter onnotreg
, ontimeout
, onbusy
, or ondnd
is applied to forward calls to a trunk configured with sip-enable-diversion=true
, SIP Server now correctly adds only one @hostname to the destination DN name in the Diversion header. Previously, SIP Server incorrectly added the @hostname twice to the destination DN name. (SIP-23346)
In a NETANN-based deployment when a race condition occurs between ApplyTreatment and SendDTMF signaling handshakes, calls no longer become stuck. Previously, if reject-call-incall was set to true
, calls to the Agent DN from which SendDTMF was sent were rejected. (SIP-23264)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.09. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports a new configuration option report-error-on-routing-end:
report-error-on-routing-end
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
If set to true
, SIP Server generates EventError with ErrorCode 453 and the Call has been disconnected
error message. This applies to the scenario where the divert-on-ringing option is set to false
, and a call routed to an agent is still in the ringing state when the caller drops the call. (SIP-23130)
Note: This release is not recommended for integration with SIP Feature Server. See known issue SIP-23463.
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If both greeting-notification=started
and record-agent-greeting=true
are configured, when the greeting is finished, SIP Server now generates EventPrivateInfo to notify the agent that the greeting is finished. Previously, in this scenario, SIP Server did not generate the greeting completion notification. (SIP-23195)
In the following scenario, if wrap-up-time=<non-zero value (0
)> and inbound-bsns-calls=true
are configured on both Routing Points (RP1 and RP2), SIP Server no longer terminates unexpectedly while generating EventEstablished:
(SIP-23015)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server provides enhanced support to SIP Feature Server for assigning calling profiles (such as dial plans) to Routing Point and Trunk Group DNs for predictive calls made on behalf of a Routing Point or Trunk Group DN. This functionality is required for consistency with the internal SIP Server dial plan. A new Application-level configuration option, enable-outbound-ext-dial-plan, supports this feature:
enable-outbound-ext-dial-plan
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next outbound scenario
Enables use of the external dial plan for outbound calls triggered by a TMakePredictiveCall request on a Trunk Group DN or RoutingPoint.
A new value, agentid
, is added to the rp-use-dial-plan configuration option and to the UseDialPlan of the Extensions attribute, for use when digit translation is not required. When set to agentid
, no dial plan is applied to the destination of TRouteCall; only an agent ID
provided by SIP Feature Server is added to the response.
This feature depends on support from a specific version of SIP Feature Server. Consult SIP Feature Server documentation for the availability of this new feature.
(SIP-22851)
This release includes the following corrections and modifications:
When the sip-filter-media option is set to video
, SIP Server now correctly forms SIP messages with SDP Media m=video
lines set to 0
(zero). Previously, SIP Server incorrectly removed the m=video
line from the SDP. (SIP-22986)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When the userdata-map-format option is set to sip-headers-encoded
,
SIP Server now encodes the user data key values of integer type. Previously,
only the string data type was encoded correctly. (SIP-23032)
SIP Server now rejects TSingleStepTransfer of the main call if the related consultation call is present. (SIP-22815)
SIP Server now distributes EventError in response to a TRouteCall request
when a routing target redirects a call to a DN, which is not available to
accept the call, by sending the 302 response. Previously, when the first
target sent a 180 response before the 302 response and the divert-on-ringing
option was set to false
, the call was redirected to an unavailable
DN and became stuck. (SIP-22539)
SIP Server no longer incorrectly releases a main call when its related
consultation call, after routing to an agent, was redirected by the 302
response to another agent that was not available to accept that call. This
issue applied when the dual-dialog-enabled option was set to
false
. (SIP-22513)
When SIP Server receives the TRouteCall request containing the OtherDN
attribute with a value of dn@host:port
, it sends the 302 response
containing the proper contact as dn@host:port
. Previously, in this
scenario, SIP Server included the invalid contact value
(dn@host:port@host:port
). (SIP-22154)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now provides a ringback tone to 1pcc calls. This behavior applies to inbound calls from external DNs. To enable a ringback tone to be unconditionally played to an external caller before the call is being placed to an agent, set the following parameters in the inbound Trunk DN:
2
true
(the default)With the sip-ring-tone-mode option set to 2
(the new value), SIP Server plays an audio ring tone only to an inbound external call, by connecting Media Server, before the call is placed to an agent.
(SIP-22462)
This release includes no corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now stops the no-answer supervision timer, when an agent answered a call but re-INVITE sent to the caller was failed. Previously, SIP Server reported the agent as NotReady with the reason ''no-answer'' because the no-answer supervision timer expired. (SIP-22767)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while processing session refresh transactions for unresponsive devices. (SIP-22900)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If an incoming trunk DN contains the sip-early-dialog-mode option set to 1
and an incoming INVITE message has no SDP and no Supported: 100rel
parameter, SIP Server now sends a 200 OK message. Previously, SIP Server incorrectly sent the 183 Session in Progress message with the SDP and waited for a PRACK that never came because of unsupported 100rel, which resulted in a stuck call. (SIP-22777)
SIP Server no longer terminates unexpectedly when a 1pcc outbound call from the shared line member is placed on hold and then an attempt to seize the line is made by the same shared line member of the shared line group. Now, in this scenario, the first call will be released and the second call will be connected. (SIP-22763)
SIP Server no longer sends a re-INVITE message to a busy destination party. Previously, when a destination party responded with a 486 Busy Here message while SIP Server had not yet received a response to a NOTIFY message, SIP Server sent re-INVITE to the busy party. The problem was that while SIP Server was processing a TSingleStepTransfer by REFER request from DN1 and sending re-INVITE to DN2 (configured with refer-enabled set to false
), SIP Server sent NOTIFY to DN1, which did not respond immediately. (SIP-22746)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When recording is enabled and the disable-media-before-greeting option is set to true
, SIP Server now correctly handles a TMakeCall request to an external DN, from the parked agent with a nailed-up connection. Previously, a ringback was not played to an agent on the outbound call if the previous inbound call was dropped by a caller before the recording was established. (SIP-22835)
SIP Server now correctly passes a 486 Busy response to the call originator after receiving the 302 Redirect message and if the 486 Busy response was the first response from the new contact to the INVITE request. (SIP-22655)
In a multi-site deployment where the same Agent Login object is configured under Switch1 on SIP Server1 and Switch2 on SIP Server2, when the Agent Login object is deleted from Switch1 explicitly, it is now deleted only in the SIP Server1 application. When a SUBSCRIBE message is received on SIP Server2 from a DN associated with the Agent Login (broadsoft-hoteling event), SIP Server2 will send NOTIFY containing AgentID associated with the DN. Previously, in this scenario, this Agent Login was also deleted from the SIP Server2 application, and for the SUBSCRIBE request to this Agent Login on Switch2, SIP Server generated NOTIFY containing no AgentID and terminated the hoteling feature. (SIP-22578)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.08. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support Genesys Cloud implementations only. (SIP-22748)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports the SIP Event Package for Registrations, in accordance with RFC 3680. A subscriber can send a subscription to SIP Server will be notified about the current state of all SIP Endpoints configured with the contact option, for a particular SIP Server switch. To support this feature, configure the following options: subscription-event-allowed and subscription-max-body-size. It is impossible to subscribe for a particular DN; a single subscription will provide notification for all DNs configured with the contact option.
The subscription-event-allowed option includes a new valid value, reg
. For this feature, set this option to reg
(SIP Event Package for Registrations subscription) or * (asterisk), which allows all subscriptions.
subscription-max-body-size
Default Value: 14336
Valid Values: 0–500000
Changes Take Effect: Immediately
Defines the maximum size of the NOTIFY XML body (in bytes) within the SUBSCRIBE dialog. If it is set to 0
(zero), the message body can be any size. The zero value can be used for TCP transport but is not recommended for UDP. For bulk notification, SIP Server sends more than one NOTIFY, so adjust the size accordingly.
(SIP-22250)
This release includes the following corrections and modifications:
When an agent changes the state from Not Ready with a Reason Code to Not Ready with no Reason Code by issuing the TAgentNotReady request, and when SIP Server receives the re-subscription request from an IP phone, SIP Server now properly synchronizes the Not Ready state with no Reason Code between the IP phone and the Agent desktop. Previously, SIP Server added a Reason Code from the previous state in the NOTIFY message for re-subscription. (SIP-22657)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When operating in HA mode, if the TApplyTreatment request for a music treatment is initiated with the duration parameter set to the maximum negative value (-2147483648
), SIP Server no longer terminates unexpectedly while synchronizing treatment attributes with the backup server. Now, when the duration parameter contains a negative value, SIP Server returns an EventError message (Invalid attribute
). (SIP-22669)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports a new Application-level option:
unknown-gateway-reject-code
Section: TServer
Default Value: 0
Valid Values: 0–699
Changes Take Effect: Immediately
false
, the unknown-gateway-reject-code defines which SIP error code SIP Server returns when an incoming INVITE message cannot be associated with an internal device or trunk. If the value of this option is less than 400
, SIP Server uses the 404 Not Found error code (the same as prior to this release). If the value of this option is 400–699
, SIP Server returns the corresponding error code. (SIP-22696)
This release includes the following corrections and modifications:
When dn-scope is set to office
, SIP Server no longer terminates unexpectedly while clearing the call after tromboning. (SIP-22597)
When sip-enable-moh is set to true
and record is set to true
, SIP Server now correctly handles a TReleaseCall request issued during TAlternateCall request processing. Previously, SIP Server did not properly process a race-condition scenario, when a TReleaseCall request was issued in the middle of the media services connection. (SIP-22590)
When a call is waiting to be routed, if sip-enable-call-info is set to true
, sip-server-inter-trunk is set to true
, and ringing-on-route-point is set to false
, SIP Server no longer sends NOTIFY with call information. SIP Server continues sending NOTIFY after the call is connected as usual. Previously, in this scenario, immediately after the INVITE message for ISCC RouteCall, SIP Server sent NOTIFY with call information, which was rejected by the second SIP Server with the 481 error message that led to the failed call. (SIP-22577)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a multi-site Complete Transfer scenario with a call recording in progress, SIP Server no longer drops the call at the moment of the transfer completion. Previously, in this scenario, because of the race condition between SIP and ISCC messages, SIP Server sometimes dropped the call. This issue occurred when the record-moh option was set to false
. (SIP-22546)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When an agent first sends the presence NOTIFY message with status Busy, and then changes the status to Ready, by sending another presence NOTIFY message during the After Call Work (ACW) time period, SIP Server now saves the latest state and updates the state after the ACW timer expires. Previously, SIP Server ignored the agent state update that was requested during ACW. (SIP-22437)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.07. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server now supports the masking of DTMF information in the following attributes of T-Library messages of SIP Server log files: AttributeCollectedDigits, AttributeLastDigits, AttributeDTMFDigits, and AttributeTreatmentParms. To use this feature, set the hide-tlib-sensitive-data option to true
in the log-filter section. For the option description, see the Genesys Configuration Options Reference Manual. (SIP-22111)
SIP Server provides enhanced reporting of the monitoring mode in EventUserEvent messages to ICON that includes LCT parties information. (SIP-21669)
This release does not include other corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.06. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server is built with TSCP version 8.1.011.06 that corrects reporting AttributeCtrlParty in EventCallDeleted. Previously, the order of notifications and sending events was incorrect; as a result, AttributeCtrlParty was not included in EventCallDeleted. (SIP-22365)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.04. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release does not include other corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.04. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When running in multi-threaded mode (sip-link-type=3
) and if one of the SIP Server threads takes more than 15 seconds to start, SIP Server now correctly loads the DN configuration. Previously, in this scenario, SIP Server did not load the DN configuration properly; as a result, calls to those DNs were rejected. (SIP-22027)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.04. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now specifies a value for the cnonce
parameter in the Authorization header that is sent in the REGISTER message, if SIP Server receives the qop
directive in the WWW-Authenticate header field. Previously, SIP Server sent the cnonce
parameter with an empty string. (SIP-22325)
When SIP Server receives an UPDATE message (with no SDP) after sending a reliable 183 response and before receiving a PRACK message, SIP Server now responds with a 2xx message to the UPDATE message. Previously, SIP Server did not respond to this UPDATE message. (SIP-22242)
When a single-step transfer is initiated and SIP Server receives an XS response from SIP Feature Server, and if the hostname is missing in the URI of the device, SIP Server now adds a host name to the To header in the outgoing INVITE message. (SIP-22156)
SIP Server now properly clears internal storage entries related to a Routing Point when a call is diverted from the Routing Point by issuing a TSingleStepTransfer request to an external number. Previously, when some entries were not cleared, it resulted in a TRouteCall request failure when the same properties were reused in new calls. (SIP-22081)
If the divert-on-ringing option is set to false
and a TRouteCall request contains the NO_ANSWER_OVERFLOW AttributeExtension with the location specified in the format <routepoint>@<hostname>
and a call is rerouted two times after the no-answer timer expires, when the call is rerouted the third time, the call is now queued in the overflow destination and the call flow continues normally. Previously, in this scenario, the call became stuck.
Note: The configuration described in the above scenario is not recommended. To use No-Answer Supervision when the divert-on-ringing configuration option is set to false
, the values of any no-answer-overflow options (extn-no-answer-overflow, agent-no-answer-overflow, posn-no-answer-overflow, or no-answer-overflow) or the NO_ANSWER_OVERFLOW AttributeExtension must not be set (they must be empty). (SIP-21998)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.04. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server is built with TSCP version 8.1.011.04 that corrects handling of requests when two different T-Library clients send RequestGetAccessNumber with the same Reference ID for different calls. (TS-11398, SIP-22129)
SIP Server no longer terminates unexpectedly while processing a wrongly-formatted message that is passed to the management port. SIP Server now ignores such management messages that are sent with a wrong format. (SIP-22077)
The silent setup for SIP Server now works correctly on Linux platforms. Previously, the silent setup installation might fail with an error stating that the data entered for the host name was invalid, even though the host name did not contain any invalid characters. (SIP-22331)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server provides enhanced support to SIP Feature Server for assigning calling profiles (such as dial plans) to Trunk DNs.
This feature depends on support from a specific version of SIP Feature Server. Consult SIP Feature Server documentation for the availability of this new feature.
(SIP-22084)
This release includes the following corrections and modifications:
SIP Server no longer applies the dial-plan rule parameter onbusy
to members of the Hunt Group. Previously, when a call destination was the sequential Hunt Group, the onbusy
parameter was set, and a Hunt Group member declined the call, SIP Server incorrectly applied the dial-plan rule, and the call was sent to the onbusy
destination instead of the next Hunt Group member. (SIP-22235)
When the sip-enable-two-party-mute option is set to true
, SIP Server now correctly reconnects a party to the call that was put on hold by TListenDisconnected by a conference member, who then left the conference, leaving only two conference parties on the call. (SIP-22152)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When the userdata-map-format option is set to sip-headers-encoded
on a VOIP Service DN, SIP Server encodes the user data values that are passed in SIP Headers of the SIP INFO request sent to MCP, but not the user data passed in gvp:params. Previously, SIP Server encoded the user data values passed in gvp:params which was not processed by MCP. (SIP-22112)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly clears the call on a DN if the DN is set to out of service by the timeout while the initiated call from this DN is not answered yet. Previously, the SIP dialog on the DN was stuck, and if the reject-call-incall option was set to true
, no calls were distributed to this DN after it came back in service. (SIP-22065)
SIP Server no longer grows in memory while processing a GVP subscription on a voicemail VoIP Service DN. Previously, SIP Server did not delete the subscription dialog after unsubscribe, which led to accumulating undeleted dialogs and, as result, to memory growth. (SIP-22033)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now encodes both the header name and its value when the userdata-map-format option is set to sip-headers-encoded
. Previously, SIP Server encoded only the header value when this option was configured. (SIP-22047)
In a multi-site routing scenario with the extended dial plan provided by SIP Feature Server, if the override-to-on-divert option is set to true
, SIP Server now correctly sets the username of the To header as the destination DN in the INVITE message sent to the agent. Previously, in this scenario, SIP Server set the username to Anonymous
. (SIP-21959)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release contains modifications to support only Genesys Cloud implementations. (SIP-21532)
This release does not include other corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.011.02. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly handles a 100 Trying response to a re-INVITE request for the nailed-up agent connection if the event-ringing-on-100trying option is set to true
on the agent's DN. Previously, SIP Server did not process the 100 Trying response correctly, which resulted in a 500 Server Internal Error response. (SIP-21909)
SIP Server now correctly starts the OOS check for existing transports after a switchover. Previously, when transports specified by the FQDN in the contact were newly added or re-configured right before a switchover, SIP Server started the OOS check of those transports without the proper delay, which triggered the oos-force timer for those transports not yet started. (SIP-21846)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server provides enhanced support to SIP Feature Server for assigning calling profiles (such as dial plans) to VOIP Service DNs with the service-type option set to softswitch
. (SIP-21680)
This feature depends on support from a specific version of SIP Feature Server. Consult SIP Feature Server documentation for the availability of this new feature.
This release includes no corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
The primary and backup SIP Servers now exchange information about calls and trigger synchronization of missing calls to the backup SIP Server after establishing the High-Availability (HA) connection. Previously, when the backup SIP Server was restarted or the HA connection between the primary and backup SIP Servers was lost and then re-established, some calls would exist only on the primary SIP Server. Also, SIP Server did not synchronize the missing calls. See Enhanced Procedure for Upgrading SIP Server HA Pair for details. (SIP-21424)
This release includes the following corrections and modifications:
When the after-call-divert-destination option is configured and an agent releases the call after a switchover, SIP Server now correctly diverts a call to the configured DN. Previously, SIP Server did not properly synchronize the value of the after-call-divert-destination option with the backup SIP Server, as the result, after the switchover, SIP Server did not divert the call to the configured DN. (SIP-21865)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When the disable-media-before-greeting option is set to true
, and an agent makes a consultation call and alternates the call before the consultation call is established, resulting in the main call being retrieved and the consultation call being put in the held state with music-on-hold (MOH) treatment. Now, when the consultation call is established, SIP Server plays the configured greeting and does not retrieve the consultation call. Previously, SIP Server retrieved the consultation call by terminating the MOH treatment, which resulted in two active calls for the agent.
When disable-media-before-greeting is set to true
and greeting-call-type-filter is set to consult/internal/outbound
, SIP Server now disables the media only if a greeting must be played to the call according to the greeting-call-type-filter option setting. Previously, SIP Server disabled the media even though a greeting was not to be played for the call. (SIP-21506)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When SIP Server receives a NOTIFY request indicating that a Media Server that started recording is out of service and that a call with recording is placed on hold, SIP Server now correctly connects an alternative Media Server to the call to continue recording. Previously, in this scenario, no alternative Media Server was connected and recording stopped. (SIP-21683)
In a standalone deployment with more than 10,000 DNs configured with the use-register-for-service-state option set to true
or in a Business Continuity deployment with more than 10,000 DNs, SIP Server's transition to the Primary role is optimized to not exceed a few hundred milliseconds. Previously, in these configurations, SIP Server's unresponsiveness for five or more seconds during the transition to the Primary role sometimes triggered the reconnection procedure for T-Library clients. (SIP-21558)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
If the sip-enable-sdp-codec-filter option is set to true
and the audio-codecs option is configured on the inbound Trunk DN, SIP Server now correctly provides codec filtering for inbound calls. Previously, SIP Server filtered only codecs configured at an Application level. (SIP-21624)
If a Voice over IP Service device goes out of service and then back in service, the backup SIP Server no longer attempts to re-activate GVP subscription. Previously, SIP Server, while in backup mode, tried to select an active transport for future subscription, which led to repetitive log messages. (SIP-21492)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while processing TApplyTreatment requests containing Reference ID with more than 9 decimal digits. This issue occurred only on the Solaris operating system. (SIP-21592)
When an agent issues a second TSingleStepConference request to a Routing Point after the first request is routed to an external destination, SIP Server now correctly adds AttributeCallState as 2 (CallStateConferenced) in the EventPartyAdded message. Previously, in this scenario, SIP Server added AttributeCallState as 0 (CallStateOk) in the EventPartyAdded message. (SIP-21513)
When the divert-on-ringing option is set to false
, and an agent issues a TSingleStepConference request to a Routing Point, SIP Server now correctly terminates an external call leg if a TDeleteFromConference request is issued on that Routing Point. Previously, in this scenario, SIP Server did not terminate that call leg if the TDeleteFromConference request was issued before the external number, to which the call was routed, answered the call. (SIP-21484)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
The functionality of the msml-oos-recover-enabled configuration option, introduced in release 8.1.102.36, is extended and now includes support for music services. This is the updated option description:
msml-oos-recover-enabled
Section: TServer
, Application level
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
This option determines how SIP Server handles the scenario in which it detects a Voice over IP Service DN (service-type=msml
) as out of service. If set to true
, SIP Server re-connects the applied treatment, conference services, or music services—music on hold or music in queue—with an alternate Voice over IP Service DN (service-type=msml
). If set to false
, SIP Server does not re-connect the treatment, conference, or music services. This option takes effect only if Active OOS Detection is enabled.
(SIP-21393, SIP-21394)
In an HA deployment using the IP Address Takeover method, SIP Server can now establish a secure connection to LCA that is running on the host with the second SIP Server in an HA pair. Use the lca-upgrade configuration option to configure secure data exchange using TLS on connections between LCA and SIP Server. This option is configured on the Host computer on which LCA is running, and where the certificate information is available. To establish a secure connection to the remote LCA, SIP Server reads/checks the following configuration options:
1
CERTIFICATE_VALUE
>
CERTIFICATE_VALUE
>
SIP Server generates the following messages when a secure connection to the remote LCA is established or lost:
52058|STANDARD|GCTI_REMOTE_LCA_SECURE_CONNECTED|Secure connection to remote LCA is established.
52057|STANDARD|GCTI_REMOTE_LCA_SECURE_CONNECTION_FAILED|Fail to establish secure connection to remote LCA.
In addition, the following new options are added to this feature:
remote-lca-addp-timeout
Setting: [TServer] section, Application level
Default Value: 9
Valid Values: 5-3600
Changes Take Effect: When a connection to the remote LCA is established
Specifies the time interval, in seconds, that an HA SIP Server waits for a response from the remote LCA after sending a polling signal. This option applies only if the remote-lca-protocol option is set to addp
.
Note: This option is reserved by Genesys Engineering. Use it only when requested by Genesys Customer Care.
remote-lca-protocol
Setting: [TServer] section, Application level
Default Value: addp
Valid Values:
simple
—regular connection
addp
— activates the Advanced Disconnect Detection Protocol
Changes Take Effect: When a connection to the remote LCA is established
Specifies the name of the method used to detect connection failures.
Note: This option is reserved by Genesys Engineering. Use it only when requested by Genesys Customer Care.
(SIP-21262)
This release includes the following corrections and modifications:
SIP Server no longer adds mailbox info to the From header in INVITE messages that it sends in routing scenarios triggered by a TRouteCall request containing the gvm_mailbox extension key. Previously, SIP Server incorrectly added the data to the From header, which led to an incorrect identification of the voice mail originator. (SIP-21502)
If, during a 1pcc single-step transfer, a destination responds with a 407 Proxy Authentication Required message, SIP Server now sends a NOTIFY message only when the authorization negotiation is completed. Previously, in this scenario, SIP Server sent a NOTIFY message to the transferee with the 407 response included in the message body, which resulted in a dropped call. (SIP-21481)
When the TCP is configured between SIP Server and SIP Proxy, SIP Server now selects the correct SIP Proxy that is in the service state to send a re-INVITE message. Previously, if one SIP Proxy was out of service, SIP Server sometimes incorrectly selected that out-of-service SIP Proxy. (SIP-21403)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now rejects TCompleteTransfer or TCompleteConference requests with EventError when a preview interaction is in progress for any of the consultation call parties. Previously, when SIP Server received TCompleteTransfer, the call was established with the consultation call destination, even though the preview interaction was in progress on the destination, and as a result, SIP Server terminated unexpectedly when the preview interaction timer expired after the transfer was completed. (SIP-21445)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
true
in the [log] section of the SIP Server Application. See Masking sensitive data in SIP messages for details. (SIP-21018)
This release includes the following corrections and modifications:
When SIP Server is configured for Active Out-of-Service Detection, and during OPTIONS/200 OK exchange messages the time is shifted backward for some reason, a SIP Server switchover no longer occurs. Previously, in this scenario, SIP Server reported the SelfMonitoring: SIP traffic timeout detected
message according to timestamp calculations that led to a SIP Server switchover. (SIP-21181)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports the SRV FQDN—FQDN resolving to SRV records—received in the Contact or Record-Route headers of a SIP message. SIP Server also supports the SRV FQDN in the existing contact configuration option on a Trunk DN. This feature also provides new configuration options: sip-enable-x-genesys-route and sip-disable-via-srv. See SRV address support in Contact and Record-Route headers for details. (SIP-20688)
In a SIP Server deployment with the Unified OpenScape Voice platform, you can use the new sip-enhance-diversion configuration option to support call forwarding from a back office phone to an agent phone, without creating an extra T-Library call in SIP Server. See Handling Call Forwarding Loop for details. (SIP-21007)
This release includes the following corrections and modifications:
SIP Server now correctly sends re-SUBSCRIBE messages to DNs located behind the softswitch after the second switchover. Previously, after the second switchover, SIP Server, running again in the Primary mode, sometimes did not send re-SUBSCRIBE messages to DNs located behind the softswitch. (SIP-21279)
If a trunk DN subscribed to presence goes out of service, SIP Server now selects the backup contact specified in the contacts-backup option on that trunk DN and sends a SUBSCRIBE message to the backup contact. Previously, when the subscription expired, SIP Server incorrectly sent the SUBSCRIBE message to the out-of-service trunk DN. (SIP-21271)
SIP Server now parks an agent with the nailed-up connection with Media Server even though it receives an unexpected UPDATE request (without the SDP, and between 18x and 200 OK messages) from an agent leg. Previously, in this scenario, the nailed-up connection was not established. (SIP-21120)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the following new configuration option:
trunk-stats-enabled
Setting: TServer
section, Application level
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
This option enables (set to true
) or disables (set to false
) calculation of trunk statistics and capacity group statistics.
Notes:
false
does not reset trunk and capacity group statistics; it only stops SIP Server from continuing to calculate them.true
without restarting SIP Server might result in incorrect call statistics and peak call statistics for trunks and capacity groups.(SIP-21305)
The description of the msml-oos-recover-enabled
configuration option, introduced in release 8.1.102.36, is updated as follows:
msml-oos-recover-enabled
Section: TServer
, Application level
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
This option determines how SIP Server handles the scenario in which it detects a Voice over IP Service DN (service-type=msml
) as out of service. If set to true
, SIP Server re-connects the applied treatment or conference services with an alternate Voice over IP Service DN (service-type=msml
). If set to false
, SIP Server does not re-connect the treatment or conference services. This option takes effect only if Active OOS Detection is enabled.
(SIP-21014)
This release includes the following corrections and modifications:
When establishing a nailed-up connection with an agent (connect-nailedup-on-login=gcti::park
), SIP Server now includes in the Contact header the username part of the From header when sending an INVITE message to the agent. Previously, SIP Server included gcti::park
as the username part in the Contact header, which resulted in rejection of the INVITE message. (SIP-21251)
SIP Server can now use the following new option, sip-allow-update-method
, to include UPDATE request support in the Allow header of 200 OK and ACK messages in scenarios in which the UPDATE request is used for non-SDP updates. Previously, SIP Server did not include UPDATE request support in 200 OK and ACK messages, which forced some SIP endpoints to use the re-INVITE method for non-SDP updates as well.
sip-allow-update-method
Section: TServer
, Application level
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
When set to true
, SIP Server includes the UPDATE method in the Allow header of 200 OK and ACK messages.
(SIP-21238)
In an early media scenario, SIP Server now ignores the 100 Trying response received on an UPDATE request and waits for a final response to the UPDATE request. Previously, SIP Server considered 100 Trying as the final response and sent an ACK message to the peer party, which resulted in the established dialog instead of remaining in the early media state. (SIP-21285)
SIP Server no longer terminates unexpectedly if the trunk configuration update occurred when Operational Statistics is outlined for the same trunk to the log file or for the HTTP statistics response, or if the in-service status of the trunk was changed when Operational Statistics outlined for this trunk. (SIP-21230)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support only Genesys Cloud implementations. (SIP-21261)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.94. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support only Genesys Cloud implementations. (SIP-21253, SIP-2095)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the following new configuration options:
find-outbound-msml-by-location
Setting: TServer
section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true
, SIP Server selects an MSML service for outbound calls based strictly on a call's geo-location. This option applies only to a call initiated by the TMakePredictiveCall request on behalf of the Routing Point DN. When set to false
, this feature is disabled.
hide-msml-location
Setting: TServer
section, Application level
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true
, SIP Server does not include X-Genesys-geo-location
and X-Genesys-strict-location
headers in INVITE messages that it sends to GVP. Removal of these headers allows SIP Server to stay in control of geo-location selection. This configuration option can be used when all MCPs controlled by Resource Manager are deployed at the same location and Resource Manager does not need to consider geo-location in the MCP selection process.
When set to false
, this feature is disabled.
(SIP-20352)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the new DN-level configuration option, sip-response-msml-oos
, to specify the SIP response code that SIP Server sends in response to an incoming INVITE message.
sip-response-msml-oos
Setting: TServer, DN level
Default Value: An empty string
Valid Values: Valid SIP response code between 400 and 699, inclusive
Changes Take Effect: For the next call
Specifies the SIP response code that SIP Server sends in response to an incoming INVITE. This option takes effect only for inbound calls received when the MSML DN is out of service. It is supported on Trunk DNs only. It must be set on the inbound trunk and applies to calls for which this trunk is used as an origination device. If the option is not set, or set to an invalid value, this feature is disabled.
The sip-response-msml-oos
option can be configured with the existing DN-level option sip-error-conversion
, when the MSML service is available but a response to an INVITE requires a SIP response code. For example, if sip-response-msml-oos = 503
, sip-error-conversion = 404=603
, and a call is made to an unknown DN, SIP Server will respond to an incoming INVITE with the 603 SIP message.
(SIP-20634)
This release includes the following corrections and modifications:
SIP Server no longer initiates unnecessary recording while establishing a nailed-up connection when an agent logs in in accordance with the connect-nailedup-on-login
option setting. Previously, if an agent DN was configured with the record
option set to true
, recording was started when the nailed-up connection was being established. (SIP-21091)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
If the after-call-divert-destination
option setting instructs SIP Server to divert a call to a Routing Point for a survey after an interaction with an agent ends, SIP Server now properly releases the call when the survey ends. Previously, in this scenario, SIP Server did not release the call. (SIP-21122)
SIP Server now correctly plays the ring tone via Media Server when the sip-ring-tone-mode
option is set to 1
, and the first-received 180 Ringing message from a called party is without the SDP. Previously, in this scenario, when SIP Server was establishing a dialog with Media Server, another 180 Ringing message with the SDP arrived from the called party, and SIP Server erroneously started connecting an agent to the second audio dialog, which resulted in no audio for the agent. (SIP-21057)
When an agent with the same login credentials tries to log in from two different phones (each phone sends a SUBSCRIBE message), SIP Server now rejects the second login attempt by sending a NOTIFY message with AgentLoggedOffEvent
in the message body. The 3pcc RequestAgentLogin is rejected by the EventError message. Previously, in this scenario, SIP Server did not reject the second login with the same agent credentials. (SIP-18710)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a supervision call with the agent
scope, when a monitored agent transfers the call to a Routing Point, followed by a treatment application and routing to a new agent, and when the sip-treatments-continuous
option was set to true
, SIP Server now correctly connects the caller with the new agent. Previously, in this scenario, SIP Server did not connect the caller with the agent, which resulted in no audio between them. (SIP-20890)
SIP Server no longer grows in memory while merging main and consultation calls initiated on different SIP Server sites.
A new Application-level configuration option, stale-call-tout
, is introduced in the extrouter
section.
stale-call-tout
Section: extrouter
Default Value: 0
(zero)
Valid Values: 0
, and from 1 min
to 24 hr
. See the ''Timeout Value Format'' in the SIP Server Deployment Guide.
For example: stale-call-tout = 5 min
Changes Take Effect: After SIP Server restart
Specifies the time interval that SIP Server waits for a call to be updated from its last update. After this time elapses, if no new events about the call are received, SIP Server clears this call as a stale transit ISCC call by deleting the call information from memory unconditionally. The default value of 0
disables the stale calls cleanup. (SIP-19620)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
X-Genesys-sipsAppName
header of the INVITE message to GVP/Media Server. This metadata—along with ANI, CallUUID, and DNIS—is used by GVP to get information about the call for a proper retrieval of recording files. The sip-enable-ivr-metadata
configuration option enables this feature. The option can be set at the Application and DN levels, in the TServer
section. See Metadata Support for IVR Recording for details. (SIP-20602)
This release includes the following corrections and modifications:
SIP Server now correctly processes voicemail messages when both group and personal voicemail boxes are configured in SIP Feature Server. Previously, a group voicemail box message was not played if a personal voicemail box was configured. (SIP-20982)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server now correctly generates log event 52053 to indicate alternate routing when URS becomes non-operational or unresponsive. Previously, in this scenario, SIP Server did not generate that log event. (SIP-20902)
SIP Server now recovers recording correctly in the scenario when INFO for the end of the greeting is received when the recording of the call is in process of starting. Previously, in this race-condition scenario, SIP Server terminated the call. (SIP-20813)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the new configuration option:
msml-oos-recover-enabled
Section: TServer
, Application level
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
When this option is set to true
, SIP Server, on detecting a Voice over IP Service DN (service-type=msml
) as out of service, recovers the treatment services connected to that service with an alternate Voice over IP Service DN (service-type=msml
). When this option is set to false
, SIP Server does not recover the treatment service when a Voice over IP Service DN (msml) is detected as out of service. (SIP-20900)
Note: The option description was updated in release 8.1.102.47.
This release includes the following corrections and modifications:
SIP Server now includes the Max-Forwards header with a predefined value set to 70
when sending the SUBSCRIBE message to Resource Manager.
Previously, when SIP Server did not include that header, a third-party load balancer located between SIP Server and Resource Manager rejected that SUBSCRIBE with the "400 Missing Max-Forwards header field" error message. (SIP-20930)
The backup SIP Server now shuts down immediately when processing the Graceful Shutdown command and will not forcefully log agents out. Previously, when graceful shutdown was issued to the backup SIP Server, it waited for all calls to complete before shutting down; it also logged agents out. (SIP-20843)
If a caller disconnects from the call while a TCompleteConference request issued by an agent is in progress, SIP Server now terminates the main call and clears all associations between main and consultation calls after the consultation call is released. Previously, in this scenario, some call associations became stuck, which resulted in a rejected call to that agent because the reject-call-incall
option was set to true
. (SIP-20797)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When an agent disconnects from the call and the caller is routed to a post-call survey and then disconnects during the survey, SIP Server now reports AttributeCtrlParty in EventCallPartyDeleted as an agent DN, because the agent disconnected first. Previously, if a caller disconnected during a post-call survey, SIP Server reported AttributeCtrlParty as the caller DN. (SIP-20835)
SIP Server no longer terminates unexpectedly when an agent transfers a call using a MuteTransfer to an external party but the caller disconnects before a provisional response is received from that external party. (SIP-20819)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the new Application-level configuration option:
msml-enable-record-extensions
Section: TServer
Default Value: true
Valid Value: true, false
Changes Take Effect: Immediately
When this option is set to true
, SIP Server sends the recording parameters in the INFO message to Media Server while restarting recording for a particular DN for which a TPrivateService request with recording parameters was issued earlier. When this option is set to false
, SIP Server does not send recording parameters when it restarts recording. (SIP-20859)
SIP Server adds new log events that are related to SIP Feature Server operation.
52056|STANDARD|GCTI_FEATURE_SERVER_URL_TIMEOUT|Feature Server URL %s missed response timeout
This message is generated when SIP Feature Server fails to reply in a pre-set timeout.
52035|STANDARD|GCTI_FEATURE_SERVER_URL_OFFLINE|Feature Server URL %s now offline
This message is generated when SIP Feature Server fails to respond on time a pre-set amount of times.
The previous alarm can be cleared by the following message:
52036|STANDARD|GCTI_FEATURE_SERVER_URL_ONLINE|Feature Server URL %s now online
These alarms can be disabled by setting the DN option enable-oosp-alarm
to false
on a Voice over IP Service DN object for a particular Feature Server.
enable-oosp-alarm
Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
When this option is set to true
, SIP Server generates alarms 52035
and 52036
(described above). When this option is set to false
, SIP Server does not generate these alarms.
(SIP-20742)
This release includes the following corrections and modifications:
When an agent logs into a DN located behind the softswitch and recording is enabled (record=true
) for the agent in the corresponding Agent Login object, SIP Server starts recording whenever a call is established with that agent. Previously, in this scenario, SIP Server did not start recording. (SIP-20877)
When running in multi-threaded mode, in a No-Answer Supervision scenario, when the divert-on-ringing
configuration option is set to false
, SIP Server now correctly processes the new TRouteCall request after the previous TRouteRequest was responded to with an EventError message because the after-routing-timeout
expired. Previously, because of a race condition, SIP Server did not process the second TRouteCall and sent EventError for all consecutive requests. (SIP-20815)
In a No-Answer Supervision scenario, when agent-no-answer-timeout
expires, but an agent answers the call while waiting for a SIP Feature Server dial plan response on a redirection request, SIP Server now sends the call to the redirect destination and releases the agent from the call. Previously, SIP Server established the call with the agent, and also sent the call to the redirect destination, which resulted in termination of the call. (SIP-20734)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.89. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
log-reduce-cpu-threshold
configuration option in the overload
section. (SIP-19817)
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly and now sends a 488 Not Acceptable Here message when it receives a REFER with Replaces request while processing the previous INVITE with Replaces request. (SIP-20805)
SIP Server now correctly handles removal of the DN-level sip-cti-control
option set to dtmf
. Previously, SIP Server incorrectly applied the value of the removed option. (SIP-20707)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.87. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
Starting with version 8.1.102.32, the modifications SIP-19469, SIP-19718, and SIP-20208 made in previous versions have been removed and are no longer available.
SIP Server can now resolve a DNS SRV record of the Trunk Group DN contact to an IP address, regardless of the number of Trunk Group DNs configured in the Configuration Database. Previously, if there were more than 256 of these DNs configured in the Configuration Database, SIP Server sometimes did not resolve the DN contact correctly. (SIP-20795)
SIP Server now correctly handles re-INVITE messages containing multiple disabled "m=" lines (0 port) in the SDP, which arrived while SIP Server was processing the previous transaction. Previously, during the SDP negotiation, SIP Server sometimes sent a SIP 503 error message instead of the re-INVITE with the SDP. (SIP-20750)
SIP Server no longer adds the ReasonCode, previously set by an agent, in EventAgentNotReady distributed on a TAgentLogin request that is submitted without any ReasonCode in it. Previously, starting with version 8.1.102.22 (correction of SIP-20410), when an agent logged out, SIP Server added the previously used ReasonCode in EventAgentLogout and EventAgentNotReady after the agent re-logged in. (SIP-20718)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.87. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
music-on-hold
configuration option or using the music-on-hold
key in AttributeExtensions of TRouteCall. The default-music
option is now supported for Agent Logins. See Customizing Music on Hold for details. (SIP-19317)
This release includes the following corrections and modifications:
In multi-threaded mode, in a monitored call with the agent
scope and recording enabled on all parties, when the monitored agent completes the transfer and leaves the call, SIP Server now starts recording the remaining parties of the call. Previously, SIP Server did not start recording after the transfer completion. (SIP-20620)
SIP Server is built with the latest TSCP, which corrects the following issue:
SIP Server now correctly processes suspended events in scenarios where the epp-tout
option is set to a non-zero value if other suspended events are present on the same party. SIP Server reports the correct AttributePreviousConnID in EventPartyChanged on multisite transfer completions. (TS-11165, SIP-20546)
SIP Server now correctly processes a call after receiving a 400 Bad Request response to a REFER request. Previously, SIP Server did not process the call correctly when multiple TRouteCall requests were sent and a response to the REFER message was received after the after-routing-timeout timer was started, which resulted in a stuck call. (SIP-20526)
When a single-step transfer to a Routing Point fails, SIP Server now deletes this party. When the same Routing Point is used in the call again, SIP Server now handles it correctly. Previously, when a TRouteCall request was received from URS for the same Routing Point later in the call, SIP Server generated an Invalid Attribute error. (SIP-20517)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
no-response-dn
option, or released if that option is not configured. A new DN-level configuration option, sip-trying-timeout
, enables this feature. See Setting SIP INVITE timeout for individual DNs for details. (SIP-20351)
This release includes the following corrections and modifications:
When the sip-error-overflow
option is configured and an agent rejects the call, SIP Server queues the call back to a Routing Point; and when a treatment is applied on that Routing Point, the caller will be able to hear the treatment. Previously, in this scenario, the caller did not hear the treatment if the disable-media-before-greeting
option was set to true
. (SIP-20616)
SIP Server now starts other scheduled services, such as supervision, when a customer greeting is successfully established, but the agent greeting is failed because of the MSML error response received from Media Server. Previously, in this scenario, SIP Server did not start other scheduled services. (SIP-20437)
When conference and recording are serviced by the same MCP and it is detected as out of service, SIP Server tries to recover conference and recording. Even if SIP Server receives an error response from an active (running) Media Server (or the timeout expires, or there is no response to INFO) during a service recovery provided by this Media Server, SIP Server is now able to clear the recorder reservation correctly and route the next call to the same DN. Previously, in this scenario, SIP Server did not clear the recording from the DN and new calls were rejected for this DN. (SIP-20122)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server supports HTTP Live Streaming (HLS) in the following scenarios:
To use this feature, SIP Server must be integrated with MCP version 8.5.161.34 or later. See HTTP Live Streaming for details. (SIP-20702)
This release includes the following corrections and modifications:
When the default-route-point
option is set to reject=<SIP ERROR>
and the default-route-point-order
option is set to after-dial-plan
, SIP Server no longer applies the default-route-point
option setting for the call if a dial plan fits the original target and the modified target, after the dial plan was applied, matches an internal resource, and SIP Server sends the call to that resource. Previously, in this scenario, when SIP Server applied the default-route-point
option setting, it resulted in EventError and left the call stuck. (SIP-20672)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When the MCP fails to play a treatment and an RM fails, and a SIP Server switchover occurs during the treatment recovery procedure, the new primary SIP Server now completes the treatment recovery procedure without delay, sending the re-INVITE message to the in-service RM right after the time period, configured in options oos-check
and oos-force
, expires. Previously, in this scenario, SIP Server delayed sending re-INVITE for the treatment after its switchover. (SIP-20589)
After establishing an inbound call to IVR (GVP) and recording for this call is started as configured, if SIP Server receives an INFO message from the IVR while recording is in progress, it will process this INFO message properly. SIP Server updates the party data accordingly and sends the 200 OK in response. Previously, SIP Server rejected this INFO with a 488 error response. (SIP-20579)
SIP Server no longer terminates unexpectedly while processing an INVITE message received with a malformed Request-URI, for example, containing '(' brackets in the DN number. (SIP-20527)
While running in backup mode, SIP Server no longer changes the media service to the out-of-service state because of DNS resolution failure. Previously, the backup SIP Server changed the media service to the out-of-service state, which made the SIP Server HA pair out of sync, which in turn resulted in stuck calls in the backup SIP Server. (SIP-20518)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
A call is now properly returned to the Routing Point and EventError is issued in the following scenario:
sip-treatments-continuous
is set to true
and divert-on-ringing
is set to false
, and
sip-continue-treatment-on-call-reject
Section: [TServer]
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When the option is set to true
(the recommended setting), the continuous treatment that is already applied is not interrupted in both scenarios above. When the option is set to false
(the default setting to ensure backward compatibility), the continuous treatment is terminated in the scenarios above as soon as TReleaseCall or TRedirectCall is received.
(SIP-20557)
In cases where a dial plan is created with some invalid rules, SIP Server now ignores invalid dial plan rules and creates a dial plan with valid rules. Previously, if the last dial-plan-rule
was invalid, SIP Server did not ignore it, and as a result, did not use the dial plan. (SIP-20519)
When in response to the INFO message generated by the TSendDTMF request, the re-INVITE message comes before 200 OK, SIP Server now schedules the processing of the re-INVITE and processes correctly TSendDTMF. Previously, in this scenario, the re-INVITE was lost and was not passed to other leg. (SIP-20428)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
charge-type
key-value pair in AttributeExtensions to control early media with a routing strategy. See Controlling Early Media With a Routing Strategy for details. (SIP-20024)
SIP Server supports Security Pack 8.5.x and provides support of TLS version 1.2.
Starting with this release, the sip-tls-sec-protocol
configuration option introduced in 8.1.101.28 has been modified, as follows:
TServer
SSLv23
SSLv23, SSLv3, TLSv1, TLSv11
If configured, this option specifies the lowest version of TLS that SIP Server will use to send and accept secure connection requests with SIP devices. This option can be used only on UNIX operating systems with Genesys Security Pack on UNIX 8.5.100.09 or later. The option has no effect on Windows. TLS versions are as follows:
SSLv23
—The highest TLS protocol version supported by Genesys Security Pack 8.5.1. Currently, it is TLS version 1.2.SSLv3
—SSL version 3.0.TLSv1
—TLS version 1.0.TLSv11
—TLS version 1.1.If not configured or set to SSLv23
(the default), SIP Server uses the highest TLS version supported by Genesys Security Pack 8.5.1. Currently, it is TLS 1.2.
(SIP-20454)
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while processing a Media Server failure for recording in scenarios with single-dialog devices. (SIP-20493)
SIP Server no longer terminates unexpectedly while releasing a call with recording after a 1pcc single-step transfer if a transferred DN was moved to the out-of-service state during the call. (SIP-20426)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.85. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains modifications to support Genesys Cloud implementations only. (SIP-20512)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.84. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
rp-use-dial-plan
and enable-iscc-dial-plan
. See Dial Plan enhancements including support for SIP Feature Server Dial Plan for details. (SIP-19929)
This release includes the following corrections and modifications:
On receiving re-subscription from an IP phone, SIP Server now properly synchronizes the Not Ready Reason code between the IP phone and the agent desktop. Previously, SIP Server did not add the Reason code for re-subscription, which made the IP phone and the desktop out of sync. (SIP-20410)
When an agent is logged out during the preview interaction timeout, SIP Server now keeps the agent in the logged out state. Previously, in this scenario, SIP Server sometimes set that agent back into the logged in and NotReady state; after that it was not possible to set the agent into the Ready state. This issue occurred when the Application option forced-notready
was set to true
(the default).
(SIP-20395)
In Business Continuity deployments, SIP Server now sends EventAddressInfo with the correct AttributeReferenceID and the agent state to the SIP Feature Server request TQueryAddress and SIP Feature Server generates dial-plan rules accordingly. Previously, SIP Server did not include AttributeReferenceID in EventAddressInfo when sending it to SIP Feature Server, which led to a timeout exception in SIP Feature Server. (SIP-20367)
When recording and treatment are being applied to a conference call and SIP Server routes the call to an agent who does not answer and the agent that initiated the conference, which was recorded, leaves the call, SIP Server now correctly re-routes the call to deposit a voicemail. Previously, in this scenario, the TRouteCall request to the Media Server containing record
and gvm_mailbox
AtrributeExtentions was rejected with an error, because of the absence of the recorded party. (SIP-20290)
In a deployment with multiple Media Servers and the Active-Active RM pair configuration, when all RMs go down and then come back in service during a SIP Server switchover, SIP Server now correctly sets the RM DN that has oos-error-check
set to true
in the in-service state after the second switchover. Previously, SIP Server sometimes did not set the DN in the in-service state after it received the positive response for the OPTIONS request. (SIP-20104)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.83. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
sip-enable-two-party-mute
configuration option enables this functionality. See Muting/Unmuting a Party in a Conference for details. (SIP-19927)
This release includes the following corrections and modifications:
In a multi-site deployment, when a re-INVITE message for the hold contains an SDP with several "m=" lines, and a second re-INVITE for a music-on-hold SDP contains several "m=" lines (one of which is inactive), SIP Server now processes those re-INVITEs correctly and a music-on-hold treatment is played to the call on hold. Previously, in cases when the sip-hold-rfc3264
option was set to false
, SIP Server took into account rejected media streams as the hold SDP, which resulted in a failed music-on-hold treatment. (SIP-20208)
Note: Starting with version 8.1.102.32, this modification is no longer available as it did not function as expected.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.83. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
This release contains corrections to support Genesys Cloud implementations only. (SIP-20419)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.82. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
3
for the sip-replaces-mode
option. In configurations with the sip-server-inter-trunk
option set to true
and the sip-replaces-mode
option set to 3
, SIP Server uses the re-INVITE method instead of the REFER method for transfers and call routing. (SIP-20326)
This release contains corrections to support Genesys Cloud implementations only. (SIP-20330, SIP-20139)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.79. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If the no-answer action is set to notready
for an agent, SIP Server can now process or ignore presence SIP messages to change an agent state to Ready, depending on a new ignore-presence-after-nas
option setting. Previously, if the no-answer action is set to notready
for an agent, SIP Server ignored presence SIP message and did not change the agent state.
If the no-answer action is set to none
for an agent, SIP Server can now process presence SIP messages to change an agent state regardless of the ignore-presence-after-nas
option value. Previously, if the no-answer action is set to none
for an agent, SIP Server ignored presence SIP message and did not change the agent state.
ignore-presence-after-nas
Section: TServer
, Application and DN levels
Default Value: true
Valid Values: true, false
Changes Take Effect: At the next call
If set to true
, SIP Server ignores presence SIP messages if the no-answer action is set to notready
.
If set to false
, SIP Server processes presence SIP messages if the no-answer action is set to notready
.
(SIP-20238)
SIP Server now correctly clears the call on a DN in scenarios in which a TInitiateTransfer request from a DN is in progress and SIP Server receives another T-Library request for that DN. Previously, a SIP dialog remained stuck and SIP Server rejected a new call to the same DN, in accordance with the reject-call-incall=true
setting. (SIP-20153)
In this scenario:
sip-server-inter-trunk
must be set to true
on a trunk between SIP Servers. (SIP-20190)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.79. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now provides additional information for historical reporting for the environment where divert-on-ringing
is set to false
and for call flows when a call is not answered by the first agent and must be distributed from a Routing Point multiple times. See Improved presentation of multiple routing attempts in historical reporting for details.
To retain the previous SIP Server behavior (prior to version 8.1.102.13), set the Application-level enable-legacy-reporting
configuration option to true
.
enable-legacy-reporting
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: On the next call
Enables backward compatibility for reporting AttributeCallState that SIP Server distributes in EventReleased for an unanswered routing target party in single-site routing scenarios.
true
, SIP Server distributes EventReleased with AttributeCallState=22 (Redirected).false
, SIP Server distributes EventReleased with AttributeCallState=7 (NoAnswer).(SIP-20088)
This release includes the following corrections and modifications:
SIP Server now displays the number of logged on agents statistic correctly. Previously, the number of logged on agents statistic sometimes incremented (decremented) multiple times when TAgentLogin (TAgentLogout) were sent for the same DN. As a result, statistics displayed wrong values. (SIP-20269)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.79. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a Business Continuity deployment, when a multi-site call is routed to a DR peer based on external dial-plan rules, SIP Server now correctly forwards the call to a voicemail if no agent is logged in to both sites. Previously, SIP Server dropped the call without forwarding it to a voicemail which resulted in a loop where the call was sent to DR peers repeatedly. (SIP-20178)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.77. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
When a 3pcc call is established after receiving an unreliable provisional response from an agent, SIP Server now correctly increments the SDP version number in the re-INVITE message sent to the caller. Previously, SIP Server did not increment the version number in the re-INVITE, which resulted in no voice path between the caller and the agent. (SIP-20103)
In a scenario in which:
When conference, treatment, and recording services are provided by different Media Servers, and if any Media Server is detected as out of service, SIP Server now recovers only media services provided by that failed Media Server. Previously, in this scenario, SIP Server did not recover services properly, which led to an error response issued by an active (running) Media Server. See also known issue SIP-20122. (SIP-19968)
SIP Server now correctly synchronizes registration information after its role has been changed from backup to primary. Previously, in a race-condition scenario with two consecutive switchovers, registration information was not synchronized correctly. (SIP-19787)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.77. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
The default value of the keep-startup-file
configuration option is changed from false
to true
. In multi-threaded logging, this change applies only to the T-Server thread log file, and is not reflected in other logs that are generated by running threads. (SIP-19837)
SIP Server now always initializes the Operational Information thread log file with a value of all
for the verbose
option and, with one exception, no longer inherits values of the verbose
option specified in the Configuration Database. The one exception is the none
value, which disables all messages in the Operational Information log file. (SIP-19673)
SIP Server now supports the new transaction-state
configuration option in the extrouter
section. This option enables improved historical reporting of data for multisite scenarios where a call is successfully delivered to the destination site but is not answered by the target agent. See the TSCP Release Note version 8.1.010.77 for details. (SIP-19371)
SIP Server support of Cisco UCM v11.0 in both modes:
This release includes the following corrections and modifications:
For configurations in which SIP Feature Server dial plans are applied to destinations, SIP Server now correctly applies the replace-prefix
option value, configured on a trunk, and will use the new value when creating the Refer-To header. Previously, SIP Server did not replace the prefix and used the received user part in the Refer-to header. (SIP-20154)
When a consultation call with activated recording and treatment is routed from a Routing Point to an agent who does not answer, and the conference is completed while the call is still on the Routing Point, and the call is re-routed to a voicemail, SIP Server now correctly includes the agent's mailbox ID for depositing a voicemail. Previously, SIP Server sent the mailbox ID as none
in the INVITE message to the voicemail DN. (SIP-20117)
SIP Server no longer terminates unexpectedly while handling SIP responses that were received over reliable protocols. (SIP-20066)
In a multi-site call, when the monitored agent leaves the call and if there is more than one active party identified at another site using the LCTParty list (the Providing Call Participant Info feature is activated), SIP Server now preserves a supervisor (in mute
or coach
supervision mode) in the call. Previously, SIP Server removed the supervisor from the call when the monitored agent left the call, without considering the two active parties remaining at another site. (SIP-16373)
It is now possible to make a call to an extension DN with the reject-call-incall=true
setting after a switchover. Previously, the DN became stuck when TListenDisconnect was issued, causing the call to be rejected because of the reject-call-incall=true
setting. (SIP-20023)
Supported Operating Systems
New in This Release
Corrections and Modifications
This hot fix is no longer available as it did not function as expected. If you downloaded it, do not install it. Hot fix version 8.1.102.09 corrects the identified issue.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly when a DN, which is already in a conference, issues a single-step conference request to an external party and releases the call before receiving a provisional response from that external party. (SIP-20025)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
cpn-self
, cpn-dnis
, and cpn-digits-to-both-legs
. See Modifying the From Header in SIP INVITE for details. (SIP-19412)
This release includes the following corrections and modifications:
In a multi-site deployment, when a re-INVITE (for example, for a greeting) is received from another site in the middle of the blind transfer completion, SIP Server now processes the transfer properly and completes it successfully. Previously, in this scenario, SIP Server generated EventError in response to that re-INVITE. (SIP-19874)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a Business Continuity deployment, while operating in DR mode, SIP Server now properly updates the state of the Extension DN as back-in-service (EventDNBackInService) if a contact of that DN is changed back to the correct contact value. Previously, in this scenario, SIP Server did not update the DN state that remained in out-of-service state. (SIP-19821)
When a party is removed from the conference call, SIP Server no longer generates EventPartyDeleted with incorrect ThirdPartyDN attributes referring to a party that never joined the call. Previously, SIP Server incorrectly included a supervisor DN (who never joined the call) in attributes ThirdPartyDN and ThirdPartyRole as RoleObserver of EventPartyDeleted. (SIP-19813)
In a deployment with multiple Media Servers and the Active-Active RM pair configuration, when an outbound call is initiated, SIP Server now correctly selects a VOIP service in a round-robin fashion. Previously, SIP Server incorrectly selected only one VOIP service. (SIP-19665)
SIP Server now correctly sends the INVITE to the MSML VOIP DN that matches the call by geo-location. Previously, in deployments with Voice Treatment Ports devices configured with the contact = "::msml"
, SIP Server sometimes sent it to a MSML VOIP DN that did not match the call by geo-location. (SIP-20245)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
Because SIP Server does not support UPDATE requests for established dialogs, the Allow header of 200 OK and ACK messages that are sent in response to an INVITE request no longer includes the UPDATE request in the list of allowed requests. Previously, when SIP Server included the UPDATE request in the Allow header list of those messages, it caused some SIP endpoints to send an unsupported UPDATE for established dialogs. (SIP-19797)
While a recording stop operation is in progress and SIP Server receives any T-Library request for the same call, it will now queue that request and execute it even though an ongoing operation is canceled. Previously, in some race conditions, SIP Server did not execute the queued request when the ongoing operation was canceled which might lead, for example, to a call not being routed. (SIP-19764)
SIP Server now correctly processes an asynchronous INFO message with content-type = application/x-www-form-urlencoded
and attaches the UserData received in the message to the call. Previously, SIP Server ignored UserData received in such an asynchronous INFO message. (SIP-19731)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Starting with SIP Server release 8.1.102.01, you can define SIP Server's default action for setting the state of an agent who was not able to answer the routed call before the after-routing-timeout
expired. The new Application-level configuration option after-routing-timeout-action
or the AFTER_ROUTING_TIMEOUT_ACTION key in AttributeExtensions of TRouteCall enables this feature. See No-Answer Supervision: After Routing Timeout Action for details. (SIP-19320)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server no longer distributes the second EventAgentLogout following agent's desktop disconnection (the logout-on-disconnect
option was set to true
) if the first EventAgentLogout was already distributed for the out-of-service DN (the logout-on-out-of-service
was set to true
). Previously, in this scenario, SIP Server distributed two EventAgentLogout messages. (SIP-19675)
SIP Server no longer generates a NOTIFY message with a negative CSeq value as a result of a local counter overflow. (SIP-19652)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
If the Application-level option sip-save-rejected-sdp is set to true
, SIP Server does not include the rejected SDP stream from the last known SDP on one endpoint while generating an SDP answer to another endpoint. Previously, SIP Server tried to include the rejected SDP stream, which resulted in an incorrect SDP answer with double "m=" lines. (SIP-19718)
Note: Starting with version 8.1.102.32, this modification is no longer available as it did not function as expected.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.70. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When the msml-support
configuration option is set to true
, SIP Server now correctly generates EventHeld in response to a THoldCall request for a chat call. Previously, SIP Server generated EventError. (SIP-19619)
When dialing the Shared Line or routing, transferring, or conferencing a call to the Shared Line, SIP Server now plays a ringback to a caller, if it is configured. Previously, SIP Server did not connect the caller to the media service, and as a result no ringback was played. (SIP-19526)
SIP Server now properly processes TCancelReqGetAccessNumber requests. Previously, SIP Server rejected those requests with the error: Bad or missing transaction specific data
. (SIP-19775)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release also includes the following corrections and modifications:
In a Business Continuity deployment, where the dr-forward
option is set to no-agent
, when a call arrives at a DR peer site as a result of the DR forwarding procedure and a destination DN is out of service, SIP Server now correctly applies the onnotreg
dial-plan parameter. Previously, in this scenario, SIP Server did not apply the onnotreg
dial-plan parameter and dropped the call with the 603 Decline error message. (SIP-19613)
SIP Server no longer duplicates the P-Access-Network-Info
header in an outgoing SIP message. Previously, in some scenarios, if SIP Server received several messages with P-Access-Network-Info
headers during one call, it included several P-Access-Network-Info
headers in the outgoing message to the caller. (SIP-19582)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release does not include other corrections and modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly while processing a misconfigured dial-plan rule and entering into a loop. (SIP-19496)
SIP Server supports a new Application-level option:
sip-save-rejected-sdp
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
When set to true
, if a media stream is rejected in the SDP answer (by setting the corresponding port to zero), SIP Server saves its state and uses it as is (with the zero port) in subsequent modifications of the same SDP session, in accordance with RFC 3264.
The default value of false
is required to preserve the original SIP Server behavior and provide backward compatibility. In this case, SIP Server ignores the media stream rejection and tries to find a matching SDP media stream, which might result in duplicated media streams in the SDP. (SIP-19469)
Note: Starting with version 8.1.102.32, this modification is no longer available as it did not function as expected.
SIP Server now executes scheduled re-INVITEs even when a greeting or recording service fails. Previously, in a multi-site deployment, and because of a race condition, SIP Server canceled scheduled re-INVITEs when a greeting or recording failed, which resulted in one way audio. (SIP-19453)
SIP Server now correctly plays a greeting for the agent that is defined in the TRouteCall request if the previous routing failed. Previously, SIP Server did not play a greeting in this scenario, even if it was defined in AttributeExtensions of TRouteCall sent to the agent DN. (SIP-19445)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
If a call is made to an incorrectly configured sequential Hunt Group (no Hunt Group members were specified), SIP Server now correctly distributes the call to a default DN if one is configured, or drops the call. Previously, in this scenario, SIP Server terminated unexpectedly. (SIP-19556)
When SIP Server receives a re-INVITE in the middle of the Music-On-Hold (MOH) recovery and the MOH recovery fails, SIP Server now re-invites the caller and agent and processes further requests for this call. Previously, in this scenario, SIP Server did not process further requests for the call. (SIP-19522)
SIP Server no longer generates an EventNetworkCallStatus message when there is no Network Attended Transfer/Conference in place. Previously, in this scenario, SIP Server sometimes generated an EventNetworkCallStatus message. (SIP-19511)
SIP Server running in the HA backup mode now correctly releases the DN reservation while deleting a stuck SIP call. Previously, when resolving a complex scenario with a broken connection to the Media Server, the backup SIP Server did not release the DN reservation. As a result, after the SIP Server switchover, the new primary SIP Server rejected a new call to the same DN. The issue occurred when the DN-level option reject-call-incall
was set to true
. (SIP-19467)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer removes capacity
and capacity-group
configuration options from the DN configuration when a new DN is created in the configuration environment. Previously, after more than 100 modifications (such as enabling, disabling, creating, and deleting DNs), SIP Server sometimes removed capacity
and capacity-group
from the existing configured DNs when a new DN was created. (SIP-19494)
When the info-pass-through
option is configured, SIP Server now correctly passes the INFO message to the inbound call leg. Previously, when SIP Server was waiting for a 200 OK message in response to INFO that was sent to GVP, and received the INFO from GVP before the 200 OK arrived, SIP Server sometimes did not pass the INFO to the inbound call leg. (SIP-19486)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server can now correctly process called numbers containing special characters. Previously, SIP Server rejected the outbound call request that contained a called number started with an open parenthesis "(". (SIP-19402)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
host:port
. It now reuses the existing TCP connection. Previously, because of a race condition, if the connection had not been established yet, a new connection was created, which led to too many TCP connections for SIP Server. (SIP-19431)
This release does not contain other corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Some countries require that a customer who is on hold must be muted to the supervisor and agent(s) who are sharing the call.
The new Application-level configuration option monitor-party-on-hold
enables or disables that behavior. See Enable Customer-on-Hold Privacy for details. (SIP-17774)
Upgrade note: If you run 8.1.101.80 or earlier versions of SIP Server, Genesys recommends the following upgrade procedure:
sip-enable-call-info
option to true
.monitor-party-on-hold
option to false
.sip-enable-call-info-extended
option is set to true
.This release includes the following corrections and modifications:
In Business Continuity deployments, when a call arrives to a SIP endpoint which was not registered and the Dial-Plan Rule with the onnotreg
parameter was configured with the destination as a voicemail (onnotreg=gcti::voicemail
), SIP Server now correctly delivers the call (with the appropriate SIP headers) to the voicemail server. Previously, in this scenario, SIP Server delivered the call without the voicemail parameters, and as a result, the voicemail server did not correctly process the call. (SIP-19392)
In an Active-Active Resource Manager (RM) deployment with SIP Proxy, when an RM goes out of service in the middle of the call, SIP Server now correctly updates the Route header with the active RM contact, so the media service request is processed successfully. Previously, when one of the RMs went out of service, SIP Server did not update the Route header and the media service request was not processed.
The new DN-level configuration option sip-route-active-transport
must be set to true
on all DNs that are configured to point to the RM pair:
service-type=msml
service-type=voicemail
sip-route-active-transport
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
When set to true
, SIP Server updates the Route header with the active RM contact. (SIP-18870)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release includes the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server now correctly performs active Out Of Service detection of the Trunk DN if its contact is configured as an IPv6 address. Previously, SIP Server did not recognize the IPv6 address in the Trunk DN contact of Resource Manager, and as a result, no connection to Resource Manager was established. (SIP-19350)
SIP Server supports a new Application-level configuration option:
greeting-stops-no-answer-timeout
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
Set this option to true
in environments where both No-Answer Supervision and Personal Greeting functionality are configured. In this case, SIP Server stops the no-answer timer as soon as a 200 OK SIP response is received, indicating that the destination party has answered the call. SIP Server does not apply the no-answer action and no-answer overflow to the call even if they are configured in the corresponding options.
The default value of false
is required to preserve the original SIP Server behavior and provide backward compatibility. In this case, SIP Server does not stop the no-answer timer until EventEstablished is generated for the destination party. It might cause SIP Server to release the call on an agent DN in the middle of the greeting if greeting-delay-events
is set to true
. EventEstablished on the agent DN is generated only when the greeting is finished. (SIP-19345)
During TMakePredictiveCall processing, if re-INVITE is received from a called party when Call Progress Analysis (CPA) is in progress, SIP Server now correctly sends re-INVITE messages to the call parties after the CPA timeout expires. Previously, in this scenario, SIP Server did not send re-INVITE messages, which resulted in the wrong SDP negotiation. (SIP-19330)
If the conditional dial plan is applied, SIP Server now preserves originally dialed digits (the original-dialplan-digits
extension key) in T-Events for the new destination if another dial plan was not applied to the new destination. Previously, SIP Server did not include originally dialed digits in T-Events for the new destination. For backward compatibility, set the restricted Application-level option preserve-original-dialplan-digits
to false
. (SIP-19327)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.67. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now reliably provides the origination DN name and location in EventRinging. The agent desktop can use this information to collect extended data about the originating party, such as the agent name, and present it to the destination party while the phone is ringing. See Providing Origination DN Name and Location in EventRinging for details. (SIP-18826)
SIP Server now supports a new DN-level configuration option:
dr-oosp-transfer-enabled
Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: For the next call
In Business Continuity deployments, for special circumstances where an inbound call remains on the same site where it arrives and SIP Server puts itself Out of Signaling Path. This option is supported only for Trunk DNs pointing to external destinations. It must not be configured on the trunks between SIP Servers.
If set to false
on the Trunk DN from where an inbound INVITE is received, SIP Server stays in the signaling path if the call, after being processed on the Routing Point DN, is sent to the local Extension DN where the DR call forwarding procedure is applied to deliver the call to the corresponding DN on the peer SIP Server. If set to true
(the default), SIP Server puts itself Out Of Signaling Path.
(SIP-19368)
This release includes the following corrections and modifications:
SIP Server is built with TSCP version 8.1.010.67 that corrects the excessive memory consumption in SIP Server. The issue occurred in the environment where contact FQDN resolution for multiple trunks was configured in the local hosts
file. (SIP-19085)
SIP Server now correctly sets the agent's Not Ready state to After Call Work (ACW) when it receives the corresponding SUBSCRIBE message while the agent was in the Not Ready state. Previously, when the ACW button was pressed on the AudioCodes 420HD phone, SIP Server did not change the agent state to ACW. (SIP-19288)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.66. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports passing geo-location information formed by the routing strategy in the multi-part body of the outgoing INVITE message. See Sending Outgoing INVITEs with Multipart Body for details. (SIP-19076)
When running in multi-threaded mode, log options time_convert
and time_format
now apply to all related SIP Server threads. Note that the time_format
option applies to log records only; it does not apply to the debug output.
(SIP-19340)
This release includes the following corrections and modifications:
In scenarios where the main call recording is not started and a consultation call is recorded, invoking Complete Transfer/Conference triggers SIP Server to restart the recording with the correct metadata and recording file name after the transfer/conference is completed. As a result, the new recording associated with the main call is created. Previously, SIP Server did not restart the recording when the main and consultation calls were merged after transfer/conference completion, and the main call was recorded in the same audio file as the consultation call. (SIP-14514)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.66. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a multi-site environment, when a consultation call has two parties on site A, but ISCC signaling is tromboned through site B, an update of UserData no longer causes an unexpected release of the ISCC connection to site B. Previously, this condition prevented distribution of the required EventPartyChanged on the transfer completion. (SIP-19299)
An active VSP call list is no longer corrupted on the backup T-Server when a connection to the primary T-Server is lost and recovered. (SIP-19240)
When the greeting-after-merge
option is set to false
, SIP Server no longer incorrectly applies a greeting to a call that is established after a blind transfer completion. Previously, SIP Server ignored this option while processing blind transfers. (SIP-19225)
In multi-site environments with three SIP Server sites involved in a call, when the first 302 redirection response is arrived without a provisional 18x response, and while processing a subsequent 302 redirection response, SIP Server no longer generates an incorrect EventPartyAdded message for a transferred call. (SIP-19199)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly continues to distribute new calls to the Hunt Group in the following scenario:
a call arrives at the Hunt Group in which one or more members are busy with other calls; idle members do not answer the call within the hg-noanswer-timeout
interval, and busy members do not free themselves up within the hg-queue-timeout
interval; and the call is finally distributed to the default DN. Previously, in this scenario, SIP Server stopped distributing new calls to the Hunt Group. (SIP-19270)
SIP Server now correctly distributes the call to Hunt Group 2 in the following scenario: a call is distributed to Hunt Group 1 and, declined by all members of that Hunt Group, it is redirected to the default DN (which was Hunt Group 2, and also contained some members of Hunt Group 1). Previously, in this scenario, SIP Server did not distribute the call to Hunt Group 2. (SIP-19136)
When a main call is reconnected after receiving an unreliable provisional response (with the SDP) in the consultation call, SIP Server now increments the version number in the updated SDP sent to the agent.
Previously, in this scenario, SIP Server did not increment the SDP version number, which resulted in no voice path between the caller and the agent. The issue occurred with the following configuration: dual-dialog-enabled
was set to false
at the agent DN and reuse-sdp-on-reinvite
was set to true
at the consultation call destination DN. (SIP-19176)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new functionality:
This release includes the following corrections and modifications:
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In multi-site scenarios, where a call is routed from Site A to Site B using
the ISCC route
transaction type, SIP Server no longer terminates
unexpectedly while handling a router-timeout
expiration on an External
Routing Point. (SIP-19210)
In Business Continuity deployments, SIP Server can now correctly resolve where to send a request when two different sites have a Routing Point configured with the same DN number. A new Application level option manages this functionality:
resolve-internal-rp-by-host
Section: TServer
Default Value: false
Valid Values: false
, true
Changes Take Effect: For next call
Set to true
to specify that SIP Server will include host information
when resolving the internal Routing Point number.
Set to false
(the default) to enable the previous behavior in SIP
Server. (SIP-19149, SIP-19091)
SIP Server now always sends a PRACK request for a response which contains Require:100rel
.
Previously, in some race conditions, when SIP Server received a reliable response
while processing another scenario during the same call (for example, connecting
to Media Server) the PRACK request might not be sent. (SIP-19067)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Enhanced alternate routing for unresponsive URS/ORS. See Alternate Routing for Unresponsive URS/ORS for details. (SIP-18321)
Support for the SIP Feature Server Find Me Follow Me functionality, for any 1pcc and 3pcc calls where Feature Server dial plans are applied to destinations. See Find Me Follow Me for details. (SIP-13954)
Support for Shared Call Appearance in Business Continuity deployments. (SIP-18446)
This release includes the following corrections and modifications:
When SIP Server receives RequestNetworkPrivateService with a special gcti_provisioning
DN as destDN
, it now correctly sends NOTIFY to the contact specified in this T-Library request. Previously, SIP Server sent NOTIFY to the phone that was last registered to the gcti_provisioning
DN. (SIP-19126)
If dynamic recording is enabled for an agent and, after a transfer, SIP Server restarts recording for that agent, SIP Server now passes to GVP the AtributeExtensions dest
and dest2
, which were received in a TPrivateService request for the correct recorder selection. Previously, in this scenario, SIP Server did not pass those extensions to GVP. (SIP-19095)
While working in Business Continuity mode and the dr-forward
option is set to no-agent
, SIP Server now correctly forwards calls to its DR peer when there is no agent logged into the DN. Previously, when the agent phone (where calls are received) was registered to only one of the two SIP Servers in the BC deployment, the DR call forwarding was not performed. (SIP-19075)
After a successful completion of the out-of-signaling-path (OOSP) transfer, SIP Server no longer sends a re-INVITE to the trunk if it receives the recording failure message during the OOSP transfer. (SIP-19058)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now reliably informs agents of a supervisor's in-call presence in multisite and complex single-site scenarios, when the Providing Call Participant Info feature is enabled. See Informing Agents of Supervisor Presence for details. (SIP-16578)
Upgrade note: If you run SIP Server version 8.1.101.59 and later, Genesys recommends the following upgrade procedure:
sip-enable-call-info-extended
to false
in all backup instances of SIP Servers.sip-enable-call-info-extended
to true
in backup SIP Servers. The following Yealink IP phone models are certified and supported: SIP-T42G with firmware V80 (x.80.0.40), and SIP-T23G and SIP-T21P E2 with firmware 80 V80 (x.80.0.33).
Note: Shared Call Appearance is not supported with models SIP-T42G, SIP-T23G, and SIP-T21P E2.
This release includes the following corrections and modifications:
SIP Server can now clear all calls on a DN that was disabled in the configuration environment while in dialing state. Previously, a SIP dialog remained stuck and no calls could be made from that DN when it was re-enabled. Use the new sip-release-call-on-disable-dn
configuration option to manage this functionality.
sip-release-call-on-disable-dn
Section: TServer
Default Values: false
Valid Values: true, false
Changes Take Effect: At the next call
This option specifies whether SIP Server releases all calls for a DN that was disabled in the configuration environment. If set to true
, SIP Server releases both call's dialogs (T-Library and SIP) for the disabled DN. (SIP-19038)
SIP Server now correctly distinguishes a prompt completion INFO message containing msml.dialog.exit
in cases where two treatments are played one after another and msml.dialog.exit
is received for a first treatment after the second prompt is started. Previously, in this race-condition scenario, SIP Server incorrectly ended the second treatment while it was in progress. (SIP-19009)
In multi-site scenarios where a call is routed from Site A to Site B using the ISCC direct-uui
transaction type, SIP Server now correctly passes the isccid
contact URI parameter from a received 302 Moved Temporarily
response when sending a SIP message to the destination. Previously, SIP Server did not pass the isccid
parameter. (SIP-18977)
When SIP Server tries to recover a treatment after its switchover but the recovery fails because of the MSML error response from a Media Server, SIP Server now generates a TreatmentEnd to the treatment. Previously, in this scenario, SIP Server terminated unexpectedly while generating the TreatmentEnd. (SIP-18976)
SIP Server now correctly keeps the call forwarding setting on a DN. Previously, if any DN option was modified in the configuration environment, the TCallForwardSet request set on that DN was incorrectly cleared. (SIP-19044)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.64. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports Active Recording via MSML in IP Multimedia Subsystem (IMS) deployments.
The following known limitations currently apply to this feature:
record
extension key set to disable_destination
or disable_source
is not supported.(SIP-18311)
This release includes the following corrections and modifications:
SIP Server no longer responds with an EventError message to a TMonitorNextCall request if a previous intrusion attempt was unsuccessful. Previously, while running in the multi-threaded mode, SIP Server incorrectly rejected the new monitoring request. (SIP-18693)
SIP Server now connects a consultation call destination DN to the Music-on-Hold service in this two-step transfer scenario: TAlternateCall was performed before the consultation call answers and the transfer initiator is connected to the ringback treatment. Previously, in this scenario, SIP Server dropped the consultation call after it answered. The issue occurred when the dual-dialog-enabled
option was set to false
on the agent DN that initiated the transfer. (SIP-18713)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
In deployments where all SIP phones are provisioned and managed by SIP Feature Server, SIP Server is now able to provide phone contact information from the SIP devices located behind a Session Border Controller (SBC) to Feature Server and send NOTIFY messages to SIP phones whenever Feature Server starts synchronizing phone configurations. In the Genesys configuration environment, configure a DN of type Extension for each phone located behind an SBC with the sip-preserve-contact
option set to true
. (SIP-18258)
The following known limitation no longer applies to this feature. It was fixed in SIP Server version 8.1.101.75.
When SIP Server receives RequestNetworkPrivateService with a special gcti_provisioning
DN, instead of sending NOTIFY to the contact specified in the T-Library request SIP Server sends NOTIFY to the phone that was last registered to the gcti_provisioning
DN. (SIP-19126)
This release includes the following corrections and modifications:
An agent can now make calls (using TMakeCall requests) while remaining in the Do Not Disturb state. Previously, in releases 8.1.x, SIP Server did not support this operation. (SIP-18905)
SIP Server no longer terminates if RequestClearCall is executed for a call distributed to a sequential Hunt Group DN but not answered yet. Previously, in this scenario, if that Hunt Group had a large number of configured members (more than 100), SIP Server sometimes terminated unexpectedly. (SIP-18903)
SIP Server now properly processes a 302 Moved Temporarily
message from a trunk, if that message came at the moment when a ringback transaction triggered by the preceding provisional response was in progress. Previously, in this scenario, SIP Server might drop the call, leaving some call properties uncleared, and reject consecutive calls to that DN if the reject-call-incall
option set to true
. (SIP-18796)
SIP Server no longer terminates unexpectedly while handling call-completion SUBSCRIBE transactions. Previously, SIP Server did not clear SipCallCompletionSubscriberDialog
correctly after subscription expired, which led to SIP Server memory corruption. (SIP-18911)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Note: This version was first released as a Hot Fix on 06/08/15.
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
preconnect_timeout
CPD result from Genesys Media Server, SIP Server releases the call with CallState=7 (NoAnswer). Previously, SIP Server ignored this CPD result and did not report the call as not answered in T-Library messages. (SIP-18871)
SIP Server supports a new Application-level option:
default-route-point-orderTServer
before-dial-plan
before-dial-plan, after-dial-plan
When the option is not configured or set to a value of before-dial-plan
, SIP Server applies a default-route-point
rule before processing the target destination according to a dial plan. When the option is set to a value of after-dial-plan
, SIP Server first applies the dial plan and only after that applies the default-route-point
rule to the dial-plan result.
(SIP-18639)
default-route-point
:
reject=<SIP ERROR>
If default-route-point
is set to reject=<SIP ERROR>
and default-routepoint-order
is set to before-dial-plan
, SIP Server rejects a call without applying any dial plan.
If default-route-point
is set to reject=<SIP ERROR>
and default-routepoint-order
is set to after-dial-plan
, SIP Server applies a dial plan, as follows:
(SIP-18673)
This release includes the following corrections and modifications:
SIP Server no longer drops a call on a Routing Point in the following race-condition scenario:
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In a scenario where a call is routed to a DN with the Preview Interaction feature enabled, but the preview request is rejected with the status "expired," SIP Server now correctly applies a treatment to the call. Previously, in this scenario, SIP Server did not apply the treatment. (SIP-18830)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In an Active-Active Resource Manager (RM) deployment with the Trunk Group subscription enabled, if both RMs go down and a SIP Server switchover occurs after which at least one of the RMs comes back in service, SIP Server now sends the SUBSCRIBE message to the Trunk Group. SIP Server resends the subscription for the Trunk Group according to the sip-retry-timeout
configuration option. Previously, in this scenario, SIP Server did not send the SUBSCRIBE message, in order to receive notifications about the Trunk Group DN. (SIP-18745)
SIP Server no longer terminates unexpectedly if the key sdp-c-host
(in AttributeExtensions of the TRouteCall request that was sent to a parked agent with the nailed-up connection) contains an empty value. (SIP-18735)
If the after-routing-timeout
option set at the SIP Server Application to a value of less than the no-answer-timeout
option set at the DN, the backup SIP Server no longer terminates unexpectedly when after-routing-timeout
expires. (SIP-18598)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
In a scenario where a call is routed to a DN with the Preview Interaction feature enabled but the DN (agent) rejects the preview request and routing strategy applies new treatment, SIP Server now correctly applies a treatment to the call. Previously, in this scenario, SIP Server rejected the treatment request. (SIP-18705)
When running in the multi-threaded mode (sip-link-type=3
or
4
), and when two agents on the same site are monitored by two
different supervisors, and one agent calls the other agent, SIP Server now
correctly includes only one supervisor in the call. Previously, SIP Server
included both supervisors in the call. (SIP-18684)
In a remote supervision scenario, when there is no monitor-type
extension key present in a TRouteCall request issued to park a remote supervisor, SIP Server no longer incorrectly cancels the remote subscription and now invites the remote supervisor for all calls answered by the monitored agent. Previously, SIP Server sometimes canceled the remote subscription after the release of the first monitored call. (SIP-18627)
In a REFER-based multi-site scenario where an agent who rejoined the conference issues a THoldCall request, SIP Server now processes the request correctly and does not release the call. Previously, in this rare scenario, SIP Server released the call unexpectedly. (SIP-18523)
In a multi-site or Business Continuity configuration conference scenario
with four agents located on different SIP Servers, all involved parties are
now correctly reported in EventUserEvent using the LCTParty
extension key. Previously, in this scenario, SIP Server included an incorrect
list of parties in EventUserEvent. (SIP-16830, SIP-16566)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.58. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a multi-site supervision scenario, if a TMonitorNextCall request from a supervisor to monitor a remote agent is rejected, SIP Server now processes further monitoring requests issued by the same supervisor successfully. Previously, in this scenario, SIP Server rejected further monitoring requests from the same supervisor with EventError. (SIP-18594)
For the ASM dialing mode when an outbound call is merged using the bridging method, SIP Server now correctly releases all Media Server parties associated with the call. Previously, SIP Server did not release Media Server parties associated with the call, which led to excessive memory consumption. (SIP-18462)
In an outbound call scenario in the ASM dialing mode, if a called party disconnects during processing of the TMergeCall request, SIP Server now clears the outbound call and generates an EventError response to the TMergeCall request. Previously, in this scenario, SIP Server generated EventReleased to the outbound leg and an agent incorrectly remained in the ASM Engaged state. (SIP-18414)
When call recording is established and a recorded party is single-step transferred, SIP Server now correctly applies the sip-ring-tone-mode
option value to determine if the ringback tone should be played to the call. Previously, SIP Server ignored the option value configured on the Trunk DN and did not connect the call to a Media Server to play the ringback tone. (SIP-18521)
SIP Server built with TSCP 8.1.010.58 no longer terminates unexpectedly while handling several concurrent configuration change events for adding and deleting connections in the Configuration Layer between server applications—for example, during restoration of the connection between Configuration Server and Configuration Server Proxy. (SIP-18375)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.56. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Geo-location for MSML-based Services: strict matching enhancement. SIP Server supports strict geo-location matching for MSML-based services by ensuring that a call with a particular geo-location is served only by an MSML service within the same geo-location or by an MSML service within the alternate location (if configured). See Geo-location for MSML-based Services: Strict Matching for details. (SIP-17476)
Geo-location support by GVP enhancement. Previously, SIP Server passed geo-location data to Resource Manager only when Genesys Media Server was configured as a Voice over IP Service (VOIP) DN. Now, SIP Server also passes geo-location data to Resource Manager when Genesys Media Server is configured as a Trunk, Trunk Group, Voice Treatment Port DN, or a Voicemail VOIP DN. See Geo-location Support by GVP for details. (SIP-17407)
Note: In Active-Active Resource Manager deployments:
::msml
is configured in the Voicemail VOIP Service DN, SIP Server selects the MSML VOIP Service based on geo-location configured in the Voicemail VOIP Service DN.This release includes the following corrections and modifications:
When the disable-media-before-greeting
configuration option is set to true
at the Application level,
SIP Server now correctly connects the caller to the voicemail if a greeting is present in the call and a destination DN does not answer the call.
Previously, in this scenario, SIP Server connected the caller in hold mode to the voicemail, so that caller could not hear the prompts of the voicemail.
(SIP-18393)
In a scenario where a call from an external device is forwarded on no-answer to a Routing Point specified in the ontimeout
dial-plan parameter, SIP Server now correctly reports CallStateOK in EventQueued. Previously, in this scenario, SIP Server reported CallStateForwarded. (SIP-18398)
SIP Server no longer terminates unexpectedly while reading erroneously configured Voice over IP service DN with service-type=cos
. (SIP-18481)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.56. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server can now add a username to the Contact header of SIP responses, taking it from the incoming INVITE request URI. A new Application-level configuration option, sip-add-local-contact-user
, enables this feature.
sip-add-local-contact-user
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true
, SIP Server takes the username from the URI of an incoming INVITE request and adds it to the local contact of the created SIP dialog. As a result, the SIP response that establishes the dialog (such as a 2xx to INVITE) as well as consecutive responses and new requests within the dialog from SIP Server will have that username inside the Contact header.
(SIP-18388)
connect-nailedup-on-login
, supports this feature. See Nailed-Up Connections for details. (SIP-17959)
This release includes the following corrections and modifications:
In multi-site environments, in a remote supervision scenario where calls from a monitored Routing Point are routed to an agent on another site by means of the route
transaction type, SIP Server now invites the correct supervisor to the call. Previously, SIP Server invited incorrect supervisors to the call. (SIP-18346)
When the divert-on-ringing
option is set to false
, SIP Server now correctly generates a new EventRouteRequest message if the call was rejected by an agent with the TRedirectCall request redirecting to the same Routing Point. Previously, in this scenario, SIP Server did not process the new routing request, which resulted in a stuck call. (SIP-18356)
In a scenario where TSingleStepConference, TInitiateTransfer, or TInitiateConference requests to a Routing Point are progress, and a TSingleStepTransfer request is issued from the same DN to the same Routing Point, SIP Server now rejects the TSingleStepTransfer request with EventError. Previously, in this scenario, SIP Server did not process those requests correctly. (SIP-18296, SIP-19568)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.56. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now properly connects remaining parties to the call when a party that is added to the conference releases the call while the two-step conference completion was in progress. Previously, in this scenario, SIP Server did not connect the remaining parties of the call. (SIP-18337)
SIP Server no longer terminates unexpectedly while processing SUBSCRIBE messages for the Call Completion feature. Previously, when a new SUBSCRIBE message was received while the previous SUBSCRIBE was in the initiated state, SIP Server tried to clear the previous transaction, which caused SIP Server to terminate unexpectedly. (SIP-18289)
SIP Server now forwards the 180 Ringing
provisional response to a caller in addition to the 181 Call is Being Forwarded
response forwarded earlier. Previously, in this scenario, SIP Server was not forwarding the 180 Ringing
without the SDP to the caller. As a result, the caller did not see the redirection information when the destination phone was ringing. (SIP-18206)
SIP Server can be upgraded from any pre-8.1.101.56 version of 8.1.1 release to version 8.1.101.59 or later without service interruption. Previously, if a pre-8.1.101.56 version was upgraded to version 8.1.101.56, .57, or .58 and SIP Servers of two different versions were running as an HA pair, agent states were synchronized incorrectly. (SIP-18399)
The restricted sip-strict-cseq-match
configuration option has been added. Set this option to false
for backward compatibility with SIP devices that are not following RFC 3261 for CSeq
implementation.
TServer
true
true, false
When set to true
, this option enables strict matching of the CSeq
header.
Warning! Use this option only when requested by Genesys Customer Care.
(SIP-18386)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.56. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server now correctly routes a call to an external destination specified in the default-dn
option when the out-of-signaling-path (OOSP) is configured on the destination trunk. Previously, in this scenario, SIP Server incorrectly terminated the call. (SIP-18248)
In a scenario where the disable-media-before-greeting
option is set to true
and the destination DN rejects or does not respond to the call and the call is returned to a Routing Point, SIP Server now correctly connects the caller with the Media Server (the active SDP) to play the treatment. Previously, in this scenario, SIP Server incorrectly connected the caller with an inactive SDP and, as a result, the caller could not hear the treatment. (SIP-18238)
In IP Address Takeover HA configurations where the sip-iptakeover-monitoring
option is set to false
, the backup SIP Server now runs the VIP (Virtual IP address) down script only after the script is fully initialized. Previously, the backup SIP Server failed to run the VIP down script during its startup when the primary SIP Server was up but the VIP down script was not fully initialized. (SIP-18133)
When running in the multi-threaded mode, the backup SIP Server no longer grows in memory if it is started when the primary SIP Server was already running for a long time with active calls and nailed-up connections in progress. Previously, under rare circumstances, the backup SIP Server sometimes grew in memory. (SIP-18031)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.56. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports Shared Call Appearance (SCA) that enables a group of SIP phones to receive inbound calls directed to a single destination (shared line); that way, any phone from this group can answer the call, barge-in to the active call, or retrieve the call placed on hold. See Shared Call Appearance for details. (SIP-13953)
This release includes the following corrections and modifications:
When an agent is logged in to the phone and the phone subscription expires, SIP Server now correctly logs the agent out. (SIP-18246)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.55. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature or functionality:
Agent Login and State Update enhancement. SIP Server now supports agent-related operations—login, logout, Ready, Not Ready—performed from the phone and provides full synchronization of agent actions between the phone and agent desktop. This feature is also supported in Business Continuity Deployments. See Agent Login and State Update on SIP Phones for details. (SIP-1745)
This release includes the following corrections and modifications:
When the enable-agentlogin-subscribe
option is configured on an agent DN, SIP Server now correctly compares the password entered by the agent on the agent phone with the password configured in the Agent Login configuration object (Advanced tab) to allow the agent to log in if the passwords match. Previously, if the password was configured on the Advanced tab of the Agent Login object, SIP Server incorrectly denied access to the agent logging in to the phone. (SIP-18163)
SIP Server no longer applies the dial plan to the gcti::record
number. Previously, SIP Server incorrectly applied the dial plan to the gcti::record
number and placed the call to a Routing Point. (SIP-18021)
SIP Server now operates properly in a high-availability deployment even if the sip-link-type
option is set to 0
in the primary SIP Server and to 3
in the backup SIP Server. Previously, after a switchover, SIP Server failed to process T-Library requests with this misconfiguration. (SIP-16635)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.55. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports a TSingleStepConference request to an external destination, which, for example, enables bringing an expert in to the conference without putting the caller on hold. The single-step conference operation can be performed from a two-way call or from a pre-existing conference. While waiting for the destination to answer the call, existing call parties will continue hearing each other. If the destination party does not respond or rejects the request, the call returns to the previous state.
Feature Limitation: In multi-site deployments, the single-step conference operation to another site via ISCC is not supported.
(SIP-17919)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.55. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly adds the Max-Forwards
header in the ACK request of the session refresh transaction. Previously, SIP Server did not add the Max-Forwards
header in the ACK request sent for the session refresh transaction. (SIP-18071)
SIP Server no longer grows in memory if there is an invalid fully-qualified domain name (FQDN) configured in the contact
option of the Trunk Group DN object. (SIP-18030)
SIP Server no longer terminates the call in a scenario in which a 183 (Session in Progress) early media provisional reliable response is received while the UPDATE in early dialog was sent by SIP Server. Previously, in this scenario, if option reuse-sdp-on-reinvite
is set to true
on the trunk, SIP Server terminated the call. (SIP-17965)
When the call-observer-with-hold
option is set to true
, SIP Server can now send either an inactive SDP (a=inactive) or black hole SDP (c=IN IP4 0.0.0.0) in the initial INVITE to a supervisor, depending on the setting of the sip-hold-rfc3264
option. Previously, SIP Server sent only a black hole SDP in the initial INVITE to the supervisor. (SIP-17929)
If, while establishing a TCP connection to a SIP Endpoint, SIP Server switches to the backup mode, it now correctly handles the closure of that TCP connection. Previously, in this scenario, these TCP connections sometimes became stale and SIP Server was not able to re-open them when switching back to the primary role. (SIP-17821)
SIP Server introduces the DN-level option enforce-rfc3455
to enable blocking the P-Called-Party-ID
header in INVITE messages for environments where SIP Server acts as a User Agent Client (UAC) and RFC 3455 is strictly enforced. Previously, SIP Server always propagated the P-Called-Party-ID
header in INVITE messages that caused the request rejection in those environments.
TServer
false
true, false
If set to true
, SIP Server does not propagate the P-Called-Party-ID
header in outgoing INVITE messages in SIP environments where RFC 3455 is strictly enforced. If set to false
, SIP Server propagates the P-Called-Party-ID
header in outgoing INVITE messages.
(SIP-17786)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.55. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
SIP Server can now forward a call to voicemail if the destination DN is busy with another call. To enable this feature:
reject-call-incall
option to true
on the destination DN.onbusy
dial-plan parameter to forward calls to a voicemail DN. (SIP-17243)
Starting with version 8.1.101.15, SIP Server supports the TEXT parameter in TApplyTreatment with the following applicable treatments:
This parameter is used to specify the URL/path of the file to be played. It is sent as an item in the PROMPT list.
URL format:
protocol://FQDN-or-IPAddress:port/path/filename
For example:
http://localhost/rMessages/dk/ATP_VMIntro.wav
Or,
file://announcement/test/WelcomeGreeting.wav
(SIP-17340)
This release includes the following corrections and modifications:
SIP Server no longer incorrectly generates error messages 06080 Mandatory configuration option not found
or 06082 Mandatory configuration option has invalid value
while processing a Switch Access Code configuration change from a non-empty value to an empty one, and vice versa. (SIP-16806)
SIP Server now releases a consultation call correctly in the following scenario:
reject-call-incall
option is set to true
on an agent DN.In the scenario where a treatment is applied to a call at one Routing Point, and then the call is routed to another Routing Point where a treatment is also applied, SIP Server now properly synchronizes the call state with the backup instance. Previously, in this scenario, SIP Server sometimes did not synchronize the call state, which resulted in a treatment failure if a switchover happened while the treatment was in progress. (SIP-17889)
While operating in Business Continuity mode, SIP Server now correctly applies the onnotreg
dial-plan parameter to DNs that have no SIP registration and no contact
configured. (SIP-17871)
SIP Server now disables the ability to perform a call pickup (in the ringing state) by a Hunt Group member in sequential ringing Hunt Group scenarios. SIP Server now rejects call pickup requests with a 403 Forbidden
SIP response. Previously, Hunt Group scenarios were processed incorrectly, causing SIP Server to terminate unexpectedly. (SIP-17863)
SIP Server now correctly processes an outbound stranded call (a call that arrives at a queue with no remaining logged-in agents) and when the stranded-on-arrival-calls-overflow
option is set to release
. Previously, in this scenario, SIP Server entered into a loop, printing aTmParty::SetCause: QueueCleared
messages, while processing a stranded call routed to an ACD queue (the default destination). (SIP-17838)
SIP Server now ignores a mismatch value of the sequence number in the CSeq
header field in NOTIFY messages within the SUBSCRIBE dialog with Resource Manager. Previously, in this scenario, SIP Server rejected requests with an incorrect CSeq
header value, which it received after a Resource Manager switchover. (SIP-17834)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.54. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
no-answer
in the AttributeExtensions or AttributeReason of the corresponding EventAgentNotReady message when an agent is placed in Not Ready state after not answering a call. (SIP-17654)
This release includes the following corrections and modifications:
SIP Server now cancels an initial client INVITE transaction after detecting a transport error in the initial server INVITE transaction. Previously, in this scenario, SIP Server did not cancel an initial client INVITE transaction but only destroyed a SIP dialog upon receiving CALLED_TRANSPORT_ERROR, which sometimes led to a stuck client INVITE transaction. (SIP-17760)
SIP Server built with TSCP 8.1.010.54 no longer grows in memory while processing multi-site (ISCC) trunk optimization scenarios in which multi-site calls are routed back to the origination site. (SIP-16775)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.53. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
AnswerClass
key with values AM
, FAX
, or SILENCE
) in UserData along with CallState of EventEstablished/EventReleased messages. This CPD result can be used by a routing strategy to make decisions for call routing to an alternate extension. (SIP-17343)
SIP Server now supports IP Multimedia Subsystem (IMS) version 13.1.
This release includes the following corrections and modifications:
When running in multi-threaded mode, SIP Server now correctly passes the message-file (.lms) information to the Log library for all threads. Previously, SIP Server logged the Std 04107 Unable to load Log Messages file 'TServer.lms', error code 30
message in all thread logs during its startup. (SIP-17651)
The restricted sip-wait-ack-timeout configuration option has been modified. It can now be configured in milliseconds. Previously, it could be only configured in seconds.
sip-wait-ack-timeoutTServer
When SIP Server processes an incoming re-INVITE request, it starts a timer to wait for the ACK message to be received for this transaction. Once the ACK arrives, SIP Server sends the re-INVITE to perform the requested operation (greeting, treatment, and so on). If the option is set to 0
(zero), the current SIP Server behavior applies.
Warning! Use this option only when requested by Genesys Customer Care.
(SIP-17411)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.53. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
disable-media-before-greeting
configuration option enables this feature. See Disabling Media Before Greeting for details. (SIP-17038)
This release includes the following corrections and modifications:
The primary SIP Server now correctly synchronizes released call data with the backup SIP Server if a call was released because of the errors in call-control operations. Previously, in this scenario, the released call was stuck in the new primary (formerly backup) SIP Server after a switchover. (SIP-17475)
If force-register
, oos-check
, and oos-force
options are set on a Trunk DN, SIP Server now sends one REGISTER message at a time to the trunk. Previously, SIP Server sent an additional REGISTER message, before the first REGISTER was canceled, to the unresponsive trunk when it detected that the trunk was back in-service. (SIP-17409)
SIP Server no longer overwrites the UserData set by URS when routing a call from a Routing Point to another Routing Point on the same SIP Server. Previously, in this scenario, SIP Server incorrectly repeated the mapping of the custom header from INVITE into AttributeUserData of the second EventRouteRequest. (SIP-17263)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.53. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports Hunt Groups with the parallel distribution strategy (simultaneous ringing) in Business Continuity deployments. See Hunt Groups for details. (SIP-15621)
(SIP-16752)
TServer
false
true, false
From
header in the INVITE message sent to the origination device. If set to true
, SIP Server includes the AttributeOtherDN value from a TMakeCall, TInitiateTransfer, or TInitiateConference 3pcc request. If set to false
, SIP Server includes the resulting digits when the dial plan is applied.
(SIP-17326)
This release includes the following corrections and modifications:
When either the display-name
option is configured at the DN level option or the DisplayName
key extension is specified in a T-Library request, and if the encoding is enabled, SIP Server now correctly encodes the display name in UTF-8 format before including it in a SIP message. Previously, in this scenario, SIP Server did not encode the display name. (SIP-17208)
When processing HTTP Digest authentication for outbound calls and when a server (SBC) indicates multiple supported quality of protection (qop) options (for example, Qop="auth,auth-int"
) in the www-authenticate
header of the Digest challenge, SIP Server now generates the Digest Response using the selected qop option value. Previously, SIP Server included an incorrect Digest Response, which resulted in failed authentication of outbound calls. (SIP-17155)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
enable-strict-location-match
configuration option enables this feature. See Geo-location for MSML-based Services: Strict Matching for details. (SIP-17163)
This release includes the following corrections and modifications:
SIP Server now correctly retransmits scheduled re-INVITE messages for session refresh purposes. Previously, in cases where the session-refresh timeout was expired while another transaction was in progress, SIP Server sometimes did not send scheduled re-INVITE messages. (SIP-17390)
SIP Server is now able to handle malformed SUBSCRIBE messages. (SIP-17308)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
MonitorMode
key in AttributeExtensions, however, it will process only MonitorMode
set to coach
. Previously, SIP Server rejected TSetMuteOff with a non-empty MonitorMode
set to a value other than coach
. (SIP-17374)
This release includes the following corrections and modifications:
If an UPDATE message arrives on an initial INVITE transaction that is in the early dialog state, SIP Server now includes the P-Asserted-Identity
header present in the UPDATE message when generating an INVITE message to a destination. Previously, SIP Server ignored the P-Asserted-Identity
header present in the UPDATE message. (SIP-17311)
When msml-support
is set to true
, and when a treatment is requested on a Routing Point, SIP Server establishes a connection with Genesys Media Server and sends an INFO message with the msml
dialog to start the treatment. If SIP Server receives the INFO message with msml.dialog.exit
in the TCP connection in response to its INFO message before 200 OK arrives, SIP Server now generates EventTreatmentApplied and EventTreatmentEnd messages. Previously, SIP Server ignored this INFO and did not generate EventTreatmentEnd. (SIP-17386)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server executes SIP-to-TLib mapping from the SIP INVITE message received in response to a REFER request that SIP Server sent to an endpoint to transfer the call to a Routing Point. Previously, in a scenario where an unattended transfer was issued to a Routing Point, SIP Server did not map user data from the received INVITE.
A new Application-level option, userdata-map-invite-after-refer
, is added.
userdata-map-invite-after-refer
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
If the option is set to true
, SIP Server executes SIP-to-TLib mapping from the SIP INVITE message received in response to a REFER request that SIP Server sent to an endpoint to transfer the request to a Routing Point. If the option is set to false
, no mapping is performed from that INVITE.
Note: If SIP-to-TLib mapping is configured for both INVITE and REFER requests and the userdata-map-invite-after-refer
option is set to true
, then in cases where an unattended transfer is triggered by a 1pcc REFER, SIP Server maps data twice. The first time SIP Server maps data from the received REFER and then it maps data from the INVITE. If the same keys must be mapped from both REFER and INVITE (for example, Call-ID), the keys from the INVITE take precedence.
(SIP-17088)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly in a scenario in which an inbound call arrives at a Routing Point, but no URS clients are connected to SIP Server, and the default-dn
Application-level option is set to a nonexistent DN. (SIP-17285)
SIP Server now correctly converts the gvp:param name
multi-byte characters to UTF-8 format in the msml body of the INFO message that is sent to MCP. Previously, SIP Server encoded characters twice and, as a result, attached the incorrect User Data to the call. (SIP-17205)
SIP Server now processes the expires
parameter in the Contact
header of the 200 OK response to the REGISTER request when the Expires
header in the 200 OK is absent. Previously, in this scenario, SIP Server ignored this parameter. (SIP-17136)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
sip-enable-tcp-keep-alive
configuration option enables the TCP keep-alive mechanism. See the Keep Alive for TCP Connections for details. (SIP-16536)
This release includes the following corrections and modifications:
When receiving an UPDATE message with the P-Asserted-Identity
header after the initial INVITE, SIP Server now correctly includes the updated P-Asserted-Identity
header in the INVITE message sent to a destination. Previously, in this scenario, SIP Server did not include the P-Asserted-Identity
header that was present in the UPDATE message while generating the INVITE. See known limitation SIP-17311. (SIP-17192)
On the Linux operating system, if sip-nic-monitoring
or tlib-nic-monitoring
is set to true
, SIP Server reports the NIC status as failed only if two consecutive monitoring requests have detected the NIC to be in failed status. All unsuccessful attempts to obtain status are ignored. Previously, SIP Server reported the NIC status as failed when the status was not detected on a second attempt (the error code was returned instead of the NIC status).
Note: A similar issue is also fixed for Virtual IP address monitoring when the sip-iptakeover-monitoring
option is set to true
. Previously, in this scenario, SIP Server reported the Virtual IP address as not present.
(SIP-17186)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server now sends the correct Request-URI for the mid-dialog request when processing strict routing in accordance with RFC 3261 requirements. Previously, when processing strict routing, SIP Server incorrectly updated the Request-URI for the mid-dialog request with a value from the Contact
header. (SIP-17051)
SIP Server supports the new configuration option session-refresh-enforced
. This option controls whether SIP Server sends the re-INVITE message when the session refresh mechanism is activated. Previously, SIP Server sent a refresh re-INVITE to all devices including those that did not support the session refresh.
TServer
true
true, false
Controls whether SIP Server activates the SIP Session timer within a SIP dialog. If set to false
, SIP Server activates the SIP Session timer only if both an initial INVITE and 200 OK response to that INVITE contains the Session-Expires header. If set to true
, SIP Server, while activating that timer, ignores the absence of the Session-Expires header in the response and starts the timer based on the header presence in the request. If an endpoint does not support the session refresh mechanism, set this option to false
. The option has an affect only when the session-refresh-interval option is set to a non-zero value. You can define this option at both the Application and the DN level. The DN-level setting takes precedence over the Application-level setting.
(SIP-16741)
When the sip-from-pass-through
option is set to true
on an inbound trunk, SIP Server now correctly passes through the SIP URI parameters in the From
header of the incoming INVITE message to the outgoing INVITE. Previously, SIP Server did not pass through SIP URI parameters. (SIP-15783)
In multi-site scenarios, SIP Server processes a TDeleteFromConference request successfully and removes the requested party from the conference. Previously, SIP Server rejected a TDeleteFromConference request with EventError if the request was submitted to remove the conference initiator, or the conferenced party present at the other site, if this party was not added by the party that submitted the request. (SIP-16797)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.52. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
SIP Server now adds MsmlResponseCode
and ResponseDescription
keys in AttributeExtensions only in EventTreatmentNotApplied messages. Previously, SIP Server incorrectly also added these keys in EventTreatmentApplied. (SIP-17078)
SIP Server no longer processes the Transfer-Type=oosp
key in AttributeExtensions of TRouteCall when an incoming multi-site call arrives via an external Routing Point. Previously, SIP Server processed Transfer-Type=oosp
and sent 302 Moved Temporarily
to the routing destination. (SIP-17027)
SIP Server now correctly drops an inbound leg of a call after receiving, in response to INVITEs, 180 Ringing
, 401 Unauthorized
, and then 481 Call/Transaction Does Not Exist
messages from the destination. Previously, in this scenario, SIP Server did not drop the inbound leg of the call. (SIP-16931)
SIP Server no longer grows in memory if the overload control module rejects a request. Previously, in this scenario, SIP Server experienced a memory leak when it was built with TSCP version 8.1.010.25 through 8.1.010.51. (TS-10777, SIP-16896)
When running in warm standby mode, SIP Server now properly registers Access Resources after a switchover. Previously, in this scenario, SIP Server did not register Access Resources, which led to failed routing transactions with the Access_Resource_Not_Registered
error message. (TS-10771, SIP-16805)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.51. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When both greeting-delay-events
and greeting-after-merge
options are set to true
and a two-step attended transfer is performed, SIP Server now correctly generates EventEstablished when a greeting after a call merge is completed. Previously, in this scenario, if a call was merged while the greeting was in progress, SIP Server did not generate EventEstablished. (SIP-16928)
The sip-error-conversion
configuration option now works correctly in HoldCall scenarios. Previously, when the sip-error-conversion
option was configured on a DN, on a softswitch DN (Voice over IP Service
DN with service-type=softswitch
), or on an Application level, and a call was placed on hold, SIP Server did not convert the error code which was received in response to the hold INVITE message. (SIP-16871)
Default values of the following configuration options are now set to 0
to disable corresponding overload control functionality:
- overload-ctrl-trequests-rate
- overload-ctrl-call-trequests-rate
- overload-ctrl-call-tupdateuserdata-requests-rate
- overload-ctrl-call-tapplytreatment-requests-rate
(SIP-16930)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.51. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
overload-ctrl-trequests-rate
option, SIP Server does not count TRegisterAddress, TUnregisterAddress, and TQueryAddress requests. (SIP-16707)
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly after receiving a re-INVITE message from the caller during a session refresh. Previously, SIP Server terminated unexpectedly while executing the re-INVITE. (SIP-16841)
For the IP Address Takeover configuration (the Application option sip-iptakeover-monitoring
is set to true
), when SIP Server is not able to open a SIP port for listening, it reports the SERVICE_UNAVAILABLE status to LCA/SCS and, as result, is promoted to a backup role, after which it correctly reports the SERVICE_RUNNING status. Previously, in this scenario, the backup SIP Server remained in status SERVICE_UNAVAILABLE. (SIP-16833)
If the charge-type
option is set to 2
in the Voice Over IP Service
DN, when the agent makes an outbound call and the called destination provides an early media, the agent and destination can now hear each other correctly. Previously, in this scenario, SIP Server incorrectly placed the agent on hold, which resulted in one-way audio. (SIP-16813)
If the Application option logout-on-disconnect
is set to true
, SIP Server no longer logs Agent A out of the DN when Agent B tries to log in on the same DN from a different T-Library client and then disconnects. Previously, in this scenario, SIP Server incorrectly logged Agent A out of the DN. (SIP-16790)
If the version of the primary SIP Server is 8.1.0.x or 8.1.1.x and the version of the backup SIP Server is 8.1.101.38, the synchronization between servers after the switchover is now processed correctly. Previously, different versions of SIP Servers were not synchronized correctly, which caused the Hot Upgrade procedure to fail.
Note: The Hot Rollback procedure, from the primary newer SIP Server version to the backup older SIP Server version, is not supported.
(SIP-16645)
If enabled, SIP Server now correctly applies Unicode support to instant messages. Previously, SIP Server did not encode instant messages and passed them through without any changes.
To enable instant messages encoding received in SIP messages from UTF-8 to a local character set in T-Library messages and vice versa, set the following options:
encoding
—Activate Unicode support.encoding-area
-—Set to chat
.A new Application-level option has been added to support this feature:
encoding-area
Section: TServer
Default Value: No default value
Valid Values: A list of areas separated by a comma (,) where encoding will be applied. Supported areas:
tlibsip
—For the Mapping SIP Headers and SDP Messages feature.chat
—For the Instant Messaging feature.Specifies the list of areas where encoding applies.
(SIP-16600)
The x-sip-log
configuration option has been modified as follows:
all
. If the x-sip-log
option is configured as an empty string, SIP Server generates the log file for the Main thread only. (SIP-16577)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.51. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Support of Windows 2012 64-bit.
Support of Cisco Gateway IOS Release 15.4(1)T1.
This release includes the following corrections and modifications:
In a scenario where the subscription-id
option is configured on both the Voice over IP Service
DN with service-type=msml
that represents an Active-Active Resource Manager (RM) pair and Trunk Group
DN with contact=::msml
, when both RMs go down, the Voice over IP Service
DN goes out of service and then comes back in service, SIP Server now subscribes (using the SUBSCRIBE message) for notifications about Voice over IP Service
and Trunk Group
DNs. Previously, in this scenario, SIP Server did not send the SUBSCRIBE message to receive notifications about the Trunk Group
DN. (SIP-16756)
SIP Server now correctly restores nailed-up connections after a switchover, if the backup SIP Server was restarted before the switchover. Previously, SIP Server did not restore nailed-up connections that were established before the switchover occurred. (SIP-16595)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.51. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server supports the ISCC Path Optimization feature introduced in T-Server Common Part version 8.1.010.51. The feature is enabled by default. It removes extra links from the Event Propagation path, while keeping the event propagation functionality intact in trunk optimization scenarios. The ISCC Path Optimization feature is controlled by the path-optimization
configuration option that must be configured in the ISCC Protocol Parameters
field of the Switch Access Code. See the TSCP RN for details.
Note: The ISCC Path Optimization feature is supported only between SIP Servers. It is not currently supported in other Genesys T-Servers.
(SIP-14722)
This release includes the following corrections and modifications:
SIP Server now correctly plays a greeting in the following scenario:
If a routing attempt fails on a Routing Point and the timeout specified by after-routing-timeout
expires, SIP Server correctly generates the EventError message, as before. However, it no longer repeats the attempt to route the call by selecting an alternate trunk for it. (SIP-16468)
SIP Server now correctly generates the Min-SE
header in accordance with RFC 4028, as follows:
Min-SE
header must not be used in responses except for those with a 422 Session Interval Too Small
response code.Min-SE
header, if present in any request or response, must not be less than 90 seconds..Min-SE
header in accordance with the RFC requirements. In the case of direct calls, SIP Server now converts the 422
response to 603
by default. (SIP-15099)
SIP Server no longer incorrectly adds the P-Asserted-Identity
header in an outbound INVITE message if an outbound Trunk DN configured with the enforce-privacy
option set to id
and the enforce-trusted
option set to false
. (SIP-16653)
SIP Server supports a new key in AttributeExtensions:
Key: busy-on-rejecttrue, false
If SIP Server receives TRouteCall with busy-on-reject=true
in AttributeExtensions and if the destination responds with the 486 Busy Here
message, it generates EventDestinationBusy for the origination party. (SIP-15900)
When an agent transfers a call to an external destination and if that destination responds with a 180 Ringing
response, SIP Server initiates a ring back to the agent. Before the media session is established for playing a ring back, if the destination sends back a 302 Moved Temporarily
response with the alternate destination, SIP Server now successfully redirects the call to the alternate destination. Previously, when the 302
response was received before the media session was established, SIP Server dropped the call.
The issue occurred with the following configuration: divert-on-ringing=false
was set on the Application level and sip-ring-tone-mode=1
was set on the agent DN. (SIP-16734)
SIP Server no longer reports license errors in the log while shutting down. (SIP-16512)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.50. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
When oos-error-check
is set for a DN of Voice over IP Service
with service type set to msml
, SIP Server now checks the
response received for the OPTIONS message. If an error response is received,
SIP Server places the DN in the out-of-service state. Previously, after
receiving any response for the OPTIONS message, SIP Server kept the DN in
the in-service state.
DN-level: oos-error-check
Section: TServer
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
Related Feature: "Active Out-of-Service Detection" in the SIP Server Deployment
Guide
Checks the response message received for the OPTIONS message sent by SIP
Server. If set to true
, if SIP Server receives any SIP error
response (for instance, 480 Temporarily Unavailable
or 503
Service Unavailable
) for the OPTIONS message, it places the DN in
the out-of-service state. If set to false
, then, if SIP Server
receives an error response for the OPTIONS message, it leaves the DN in
the in-service state. This option applies only to a DN of type Voice
over IP Service
with service-type=msml
.
Note: This option must be used together with the oos-check
option.
SIP Server supports a new restricted DN-level configuration option:
device-startup-time
. This option specifies the time period
that SIP Server waits before placing the Resource Manager (RM) in the
in-service state after receiving the first positive response for the
OPTIONS message from the RM. With this option, SIP Server no longer places
the RM immediately back into service once it receives the positive response
for OPTIONS from the RM. Instead, it waits for the time period specified
in device-startup-time
before placing the RM into service. It
can take a maximum time of device-startup-time
plus
oos-check
time for the DN to come back into service.
DN-level: device-startup-time
Section: TServer
Default Value: 5
Valid Value: 0-300
Changes Take Effect: Immediately
Specifies the time period that SIP Server waits before placing the RM in service after it receives the positive response for the OPTIONS message from that RM. This option applies only:
Voice over IP Service
with
service-type=msml
.oos-error-check
is set to true
.(SIP-6022)
SIP Server now provides additional information about the location of
call participants and reports it in the Extensions attribute of EventUserEvent
in multi-site call scenarios. This feature is enabled by setting the
sip-enable-call-info
configuration option to true
.
The following information will be added by SIP Server in EventUserEvent
messages distributed to a T-Library client to show call participants:
LCTParty<n>_location
—Provides the DN location
name; that is, the name of the switch to which this DN belongs.(SIP-16229)
This release does not include any corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.50. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections or modifications:
SIP Server now correctly parks an agent with a nailed-up connection after processing a two-step transfer to a Routing Point. Previously, in this scenario, SIP Server did not park the agent with the nailed-up connection. (SIP-16479)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.50. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
This release introduces a new SIP Server Application-level option:
sip-tls-sec-protocol
Section: TServer
Default Value: SSLv23
Valid Values: SSLv23, SSLv3, TLSv1, TLSv11
Changes Take Effect: After restart
This option specifies which handshake protocol SIP Server will use on the SIP TLS listening port and when connecting to a SIP TLS-enabled device as a client. This option can be used only on UNIX operating systems with Genesys Security Pack on 8.1.300.03 or later. The option has no effect on Windows. Protocols are specified by the option values as follows:
SSLv23
—SSL version 2.0.SSLv3
—SSL version 3.0.TLSv1
—TLS version 1.0.TLSv11
—TLS version 1.1.(SIP-16335)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.49. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following features or functionality:
hg-type
configuration option that specifies the type of Hunt Group algorithm for call delivery to Hunt Group members has been extended to support linear and circular sequential distribution strategies. See Hunt Groups for details. (SIP-15180)
This release includes the following corrections and modifications:
SIP Server now correctly schedules a re-INVITE in the following inbound-call scenarios:
In IP Address Takeover HA scenarios, SIP Server now correctly transmits a 200 OK message to a destination after a switchover if the TCP connection is used between SIP Server and an endpoint. Previously, SIP Server attempted to restore the TCP connection after the switchover and did not transmit 200 OK properly. (SIP-16375)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.48. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now accepts and properly processes a TSendDTMF request if a destination party has not answered yet, but has been connected to the caller through the early media provisional reliable response. Previously, SIP Server rejected such a request. (SIP-16290)
SIP Server now correctly parks an agent with a nailed-up connection when the second switchover occurs. Previously, SIP Server correctly parked the agent with the nailed-up connection after the first switchover but failed to park the agent with the nailed-up connection when the second switchover occurred. (SIP-16317)
SIP Server now correctly performs SIP-to-TLib data mapping when a SIP REFER request is received during the execution of the TSendDTMF request. Previously, in this scenario, SIP Server did not map SIP-to-TLib data even though it processed the SIP REFER request properly. (SIP-16299)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.48. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When operating in Business Continuity (BC) deployments, SIP Server now correctly forwards the multi-site MWI notification request TNetworkPrivateService (AttributePrivateMsgID - 6002) to its BC peer. (SIP-16228)
When both contact
and contacts-backup
options are configured on the Trunk and a single-step transfer is initiated using the OOSP method (oosp-transfer-enabled=true
), SIP Server now correctly uses the active contact in the Refer-to
header of the REFER message. Previously, SIP Server incorrectly used only the primary contact and did not use the contacts-backup. As a result, when the primary contact went down, the single-step transfer could not be completed.
(SIP-16047)
If the SIP URI in the From
header of an INVITE request contains a comma, semicolon, or question mark, SIP Server now encloses the SIP URI in angle brackets (< >) when passing the INVITE. Previously, SIP Server removed the angle brackets from the From
header. (SIP-16201)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.48. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now provides the ability to control AttributeReasons to be populated in unsolicited EventAgentReady messages generated by the After Call Work (ACW) feature, by using the Application-level acw-persistent-reasons
configuration option in the TServer
section. Previously, SIP Server always populated AttributeReasons in EventAgentReady messages.
acw-persistent-reasons
Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
If set to true
, SIP Server populates AttributeReasons in unsolicited EventAgentReady messages generated by the After Call Work (ACW) feature. If set to false
, SIP Server does not populate AttributeReasons in EventAgentReady messages.
(SIP-1952, ER# 277433536)
When running on the Linux/UNIX operating system with TLS enabled, the TLS connection between SIP Server and its clients might be unstable, and SIP Server might terminate unexpectedly if the sip-tls-cipher-list
configuration option was specified. To solve this issue, install Genesys Security Pack on UNIX version 8.1.300.01. (SIP-16071)
SIP Server no longer compares host names in a security certificate with the actual host name on a case-sensitive basis. Previously, a case-sensitive comparison was made, resulting in incorrect failures in authentication. To solve this issue, install Genesys Security Pack on UNIX version 8.1.300.01. (SIP-16182)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature or functionality:
ims-use-term-legs-for-routing
configuration option enables this feature. See IMS Integration: Routed Calls as Originating or Terminating for details. (SIP-15182)
This release does not include any corrections.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly parks an agent with a nailed-up connection in a race condition scenario where the BYE message is received from the external caller and the call is terminated during routing to the agent with the nailed-up connection. Previously, in this scenario, SIP Server did not park the agent. (SIP-15906)
After a switchover, SIP Server now correctly generates an EventUserEvent message when it receives the MWI NOTIFY from a voicemail SIP Server. Previously, in this scenario, when NOTIFY was received after the switchover, SIP Server did not generate EventUserEvent. (SIP-15812)
If a graceful shutdown command is issued while calls are in progress, a SIP Server instance running in a backup mode now correctly shuts down as soon as all calls in progress are over. Previously, in this scenario, the backup SIP Server did not shut down if some calls were in progress. (SIP-15929)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In nailed-up connection scenarios, SIP Server now correctly establishes a new parking SIP dialog, after a switchover, if a backup SIP Server was started later than the primary, after the SIP dialog was already established. Previously, SIP Server did not preserve the nailed-up connection if a dialog to the nailed-up agent had been established on the primary SIP Server before the backup SIP Server had been started. (SIP-15815)
SIP Server now rejects a TSingleStepTransfer request if the request initiator is in the dialing state. Previously, in this scenario, SIP Server did not report respective events correctly to the transfer initiator and the agent DN became stuck. (SIP-15688)
When a single-step transfer is in progress, SIP Server now acknowledges the dynamic recording request for a transferred party, immediately, but starts recording only after a destination party answers the call. Previously, in this scenario, SIP Server tried to start recording before the destination party answered the call that resulted in a failed recording attempt. (SIP-15693)
When a single-step transfer initiated using the re-INVITE method fails, SIP Server now correctly retrieves the call and restores its original state. For example, if a caller is put on hold before a single-step transfer is initiated, the call is restored back to its hold state. Previously, in this scenario, SIP Server did not retrieve the call properly. (SIP-15503)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains no new features or functionality.
This version was first released as a Hot Fix on 03/18/14.
This release includes the following corrections and modifications:
When an agent initiates a consultation call by using a TInitiateTransfer or TInitiateConference request and then, when there is no response, releases it, SIP Server now allows a second attempt to initiate the consultation call to the same destination. Previously, in this scenario, SIP Server rejected the second attempt with the Request already in progress
error message. (SIP-15795)
When running in multi-threaded mode on the Solaris operating system, SIP Server no longer becomes unresponsive if one of the threads consumes 100% of its memory allocation. (SIP-15727)
When the monitor-consult-calls
option is set to true
and if a supervisor initiates intrusion monitoring when the consultation call is in the ringing phase and the main call is on hold, then SIP Server starts monitoring of the consultation call when it is established. Previously, SIP Server started monitoring of the main call on intrusion when a consultation call was in ringing state, which sometimes negatively impacted processing of a TCompleteTransfer request. (SIP-15636)
When running in multi-threaded mode, SIP Server no longer terminates unexpectedly when a connection to a USB flash drive or SAN storage on which logging occurs is broken. To avoid this issue, set the no-memory-mapping
log option to true
.
no-memory-mapping
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: At restart
Specifies if memory-mapped files, including memory log output (with file extension .memory.log
) and snapshot files (with file extension .snapshot.log
), are disabled for file outputs. (SIP-15499)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer includes the value of the display-name
option configured on a Trunk
DN in the To
header of the INVITE message that it sends to the destination DN through this trunk. Previously, SIP Server always included the value of this option in the To
header. (SIP-15293)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now supports DN Recording Override. With this feature, call recording can be selectively disabled through a routing strategy by overriding the record
configuration option enabled on a DN. Call recording can be disabled on either the origination DN or destination DN when a routing strategy issues TRouteCall containing the record
extension key with a value of disable_source
or disable_destination
, respectively. See DN Recording Override for details. (SIP-14668)
Removal overdialed digits from DNIS. SIP Server provides the ability for internal and inbound calls coming to a Routing Point to remove overdialed digits from DNIS when the dnis-max-length
dial-plan rule parameter is specified. Overdialed digits are added to the DNIS_OVER
key of AttributeExtensions in T-Library events EventQueued and EventRouteRequest. See Dial Plan Support for Overdial for details. (SIP-15299)
Call monitoring enhancement. SIP Server now supports the scenario in which a supervisor, participating in a monitoring session, can do a call transfer when an agent leaves a monitored call and the supervisor is in the call with the customer in MuteOff mode. (SIP-15277)
SIP Server now supports the TEXT parameter in TApplyTreatment with the following applicable treatments:
This parameter is used to specify the URL/path of the file to be played. It is sent as an item in the PROMPT list.
URL format:
protocol://FQDN-or-IPAddress:port/path/filename
For example:
http://localhost/rMessages/dk/ATP_VMIntro.wav
Or,
file://announcement/test/WelcomeGreeting.wav
(SIP-17340)
This release includes the following corrections and modifications:
SIP Server no longer generates EventError when processing a TSendDTMF request in a race condition. Previously, SIP Server generated EventError instead of EventDTMFSent, when an agent interrupted the announcement by sending DTMF digits from Interaction Workspace, and Media Server reached the end of announcement and sent a SIP BYE message for the announcement leg at the same time SIP Server was attempting to re-INVITE the announcement leg after the DTMF digits were collected. (SIP-15120)
SIP Server now sets off all the timers related to a treatment (START_TIMEOUT,TOTAL_TIMEOUT,DIGIT_TIMEOUT) when the treatment PlayAnnouncement and CollectDigits is completed (that is, EventTreatmentEnd is generated or the routing destination responds with 200 OK). Previously, SIP Server did not set off the DIGIT_TIMEOUT timer and the call was dropped after receiving 200 OK from the routing destination when the timer expired. This issue applied when sip-treatments-continuous=true
and msml-support=true
. (SIP-15481)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server provides the ability to block video streams from SDP offers during the call negotiation/establishment process, so video will not be played when a call is established. The sip-filter-media
configuration option enables this feature. The option can be set at both Application and DN levels. The option setting at the DN level takes precedence over the Application-level setting. See Video Blocking for details. (SIP-14847)
SIP Server enables control of the number of outgoing and incoming calls to be handled by a specific trunk or a group of trunks in single-site deployments. SIP Server rejects the calls when trunk capacity is reached. In addition to the existing capacity
configuration option, the following options have been introduced: capacity-sip-error-code
, capacity-tlib-error-code
, and capacity-limit-inbound
. See Trunk Capacity Control for details. (SIP-14846)
SIP Server now offers the Sizing Tool to evaluate the CPU load and network traffic of SIP Server and SIP Proxy applications in your environment. Download the tool from the Genesys SIP Server Documentation web page. (SIP-14141, SIP-13948)
SIP Server now supports the init-dnis-by-ruri
configuration option.
init-dnis-by-ruri
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
If this option is set to true
, SIP Server determines what value must be reported as the DNIS attribute in T-Library messages for inbound or 1pcc calls, in the following order:
If this option is set to false
, SIP Server takes the username from the To header as a value for the DNIS attribute.
(SIP-15363)
This release includes the following corrections and modifications:
SIP Server now correctly distributes calls to Hunt Group members after their busy state has ended, that is, as soon as they become available to accept new calls from queue. Previously, those Hunt Group members sometimes remained incorrectly in the busy state if SIP Server stayed active for at least 24 days after it started. (SIP-15436)
SIP Server now supports HTTP Digest authentication for re-INVITE messages. SIP Server generates a new re-INVITE with the authentication header in response to 401 Unauthorized
or 407 Proxy Authentication
challenges to the re-INVITE. Previously, SIP Server supported the authentication procedure for only initial INVITE or REFER messages. (SIP-15171)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server, when working in single-thread mode, no longer terminates unexpectedly while processing a two-step transfer or two-step conference request from an agent who was immediately released after submitting the TInitiateTransfer or TInitiateConference requests. (SIP-15407, SIP-15302)
SIP Server now correctly encodes the filename for a treatment to UTF-8 format in the msml body of the INFO message that is sent to MCP. Previously, SIP Server did not encode the filename to UTF-8 format, which resulted in a treatment error. (SIP-15359)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When msml-support
is set to true
, and when a treatment is requested on a Routing Point, SIP Server establishes a connection with Genesys Media Server and sends an INFO message with the msml
dialog to start the treatment. If SIP Server receives the INFO message with msml.dialog.exit
in response to its INFO message before 200 OK arrives, SIP Server now generates EventTreatmentApplied and EventTreatmentEnd messages. Previously, SIP Server ignored this INFO and did not generate EventTreatmentEnd.
(SIP-14759)
SIP Server now correctly puts the value of the Expires
header and expires
parameter in the Contact
header in the 200 OK response to the REGISTER Refresh request—the value is the lesser of the expires
parameter present in the REGISTER request and the registrar-default-timeout
configuration option. Previously, SIP Server included the incorrect expires
value when generating the 200 OK response to the REGISTER Refresh request. (SIP-15348)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In IMS deployments where divert-on-ringing
is set to false
, when a call is routed to an external number, SIP Server no longer diverts the call from the Routing Point if it receives a SIP 183 Session Progress
message with an SDP offer in response to the SIP INVITE sent to the external number. Now SIP Server diverts the call only after receiving a SIP 200 OK
response to the INVITE sent to the external number. (SIP-13766)
SIP Server no longer forwards INFO messages containing Content-Type: application/dtmf-relay
if the info-pass-through
configuration option is not configured. Previously, SIP Server incorrectly forwarded these INFO messages in case of Music
or PlayApplication
treatments. (SIP-15093)
SIP Server now attaches the correct User Data to the call for both MSML- and NETANN-based services if the header in the INFO message contains the data in UTF-8 format. Previously, for a NETANN-based service, SIP Server encoded data characters twice and, as a result, attached the incorrect User Data to the call. (SIP-15003)
For Instant Messaging (IM) calls, a PlayAnnouncement
treatment can now be requested more than five times without the call being dropped. Previously, SIP Server dropped a chat call if this treatment was requested more than five times. (SIP-15021)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release also includes the following corrections and modifications:
In MSML configurations, SIP Server now properly updates AttributeUserData and AttributeReferenceId in the EventTreatmentEnd message that is generated after a switchover of SIP Servers that happened while a treatment was in progress. Previously, in this scenario, AttributeUserData and AttributeReferenceId were missing from the EventTreatmentEnd message. (SIP-14958)
If the sip-error-conversion
option is configured on both Application and DN levels, SIP Server now correctly applies the DN-level value because it takes precedence. Previously, SIP Server ignored the DN-level value. (SIP-14967)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server can now apply Dial Plan rules to outbound DNs provided in the OtherDN attribute of TMakePredictiveCall when a dial plan is assigned to the DN initiating the call. For this feature to work, a designated Dial Plan can now be assigned to DNs of type Trunk Group and Routing Point.
For TMakePredictiveCall scenarios, SIP Server supports the following parameters in the dial-plan rule:
type
(with digits
and reject
values)calltype
(with outbound
values)clir
privilege
This feature applies to outbound calls in Proactive, ASM, and Transfer modes. It is not supported for Dial Plans provided by SIP Feature Server.
(SIP-13957)
This release also includes the following corrections and modifications:
For configurations with reliable transports, SIP Server now properly
retransmits 200 OK
responses to an INVITE message if
ACK
is not received for 200 OK
. To enable this
functionality, use the new option sip-rel-200-retransmit
.
sip-rel-200-retransmit
Section: TServer
Default Value: false
Valid Values: false
, true
Changes Take Effect: For next call
Specifies if SIP Server retransmits 200 OK
in response to
an INVITE message on reliable transports. This option can be set at the
Application and/or the DN level, with the DN-level setting overriding the
value set at the Application level.
The default value of this option (false
) enables the old
behavior in SIP Server.
(SIP-14953)
SIP Server now properly handles T-Library requests that have been issued in quick succession while another call operation is in progress. Previously, these requests were sometimes not handled properly. For example, TCompleteTransfer was sometimes not executed while two consecutive requests for recording were being processed. (SIP-14909)
In MSML configurations for multi-site calls using the direct-uui
transaction type, the supervisor can now change the supervision mode for
initiated Intrusion using MuteOn/MuteOff. Previously, in this scenario,
changing the supervision mode for initiated Intrusion did not work properly.
(SIP-14833)
In HA environments, memory allocations of a certain size (512 bytes) no longer cause unlimited memory consumption in a backup SIP Server. (SIP-5652)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.46. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
When SIP Server applies a dial plan to a call, it now includes the
original-dialplan-digits
extension key that contains the
original destination number (the dial plan input) before the dial plan is
applied. If a call scenario contains multiple consecutive steps (for example,
an inbound call to a Routing Point, routing to an agent, and a single-step
transfer to the other agent), then an original dialed number is defined for
each call step. For example, one dialed number is defined for an inbound
call, another for routing, and a third one for a single-step transfer.
If a destination DN is a Routing Point, then the original-dialplan-digits
extension key is passed in EventQueued and EventRouteRequest messages. If a
call is made to the ACD Position DN, then a new extension key is added to
EventQueued. If a call is made to the Extension DN, then a new extension key
is added to EventRinging.
If the dial plan is not applied to the call, original-dialplan-digits
will not be added.
(SIP-14291)
For MSML-based recording, SIP Server now supports continuous recording for conference calls if the party (where the record was initiated) is dropped from the conference, and any party remaining on the call requests the recording (by DN configuration, routing strategy, or T-Library client). Recording ends when no more recording-enabled parties remaining. Recording can be stopped before that by a respective T-Library request. (SIP-14290)
SIP Server now supports 3pcc and 1pcc blind transfer operations when a
complete transfer operation is performed while the transfer destination party
is in the alerting state. To enable this feature, set the
blind-transfer-enabled
option to true
at either the
Application level, or at the DN level of the transfer destination.
(SIP-13956)
SIP Server can now provide the ability to record a call without recording a music-on-hold treatment when a call is placed on hold. SIP Server sends corresponding MSML information in INFO messages to Genesys Media Server to pause the recording (gvp:recorder state="pause") when the call is placed on hold and to resume the recording (gvp:recorder state="start") when the call is retrieved.
This functionality also applies to call transfers: the recording is paused when a transfer is initiated, and resumed when the transfer is completed.
When several agents are involved in the call and the call is placed on hold, SIP Server pauses the recording at the first invocation of the hold operation and resumes the recording at the first invocation of the retrieve call operation.
If SIP Server receives an error message from Genesys Media Server when pausing or resuming the recording, it will not resubmit the request. Recording will be left in the previous state.
In multi-site deployments, where the recording and music-on-hold treatment might happen on different SIP Servers, the SIP Servers will use an EventNetworkPrivateInfo message containing the AttributePrivateMsgID to pass the recording control from one SIP Server to other SIP Server. The value of the AttributePrivateMsgID indicates the recording state:
Note: This feature is supported in multi-site deployments (ISCC
calls) only when event-propagation
is set to list
in the extrouter
section of the SIP Server application.
To enable this feature, use the new Application-level option
record-moh
and set it to false
.
record-moh
Section: TServer
Default Value: true
Valid Values: true
, false
Changes Take Effect: Immediately
Specifies whether the music-on-hold treatment is recorded during call
recording when the call is placed on hold. If set to true
,
the music-on-hold treatment is always recorded during call recording.
If set to false
, SIP Server pauses the recording when the
call is placed on hold and the music-on-hold treatment will not be
recorded. SIP Server resumes the recording when the call is retrieved.
Note: This option applies only if sip-enable-moh
is
set to true
and the MSML environment is used for recording.
(SIP-13595)
This version was first released as a Hot Fix on 11/15/13.
This release also includes the following corrections and modifications:
For MSML-based recording, SIP Server now only sends additional metadata in
a SIP INFO message (create conference and record) when the
msml-record-metadata-support
option is set to true
in the TServer
section at the Application level. Previously,
metadata was distributed when MSML-based recording was initiated.
msml-record-metadata-support
Section: TServer
Default Value: false
Valid Values: true
, false
Changes Take Effect: Immediately
Specifies whether SIP Server, while starting call recording, sends
additional metadata in the INFO message to Genesys Media Server. If set to
false
(the default), SIP Server does not include additional
metadata in the INFO message. If set to true
, SIP Server sends
additional metadata to the Genesys Media Server for use in Genesys Media
Server file-based call recording. (SIP-14839)
SIP Server now correctly handles a re-INVITE message containing an SDP offer
with the inactive
attribute during a pending REFER transaction
by responding with a 200 OK
message containing an SDP answer
with the inactive
attribute. Previously, SIP Server responded
with a 200 OK
message containing the last-known SDP answer.
(SIP-14827)
SIP Server now properly generates EventTreatmentNotApplied for the Routing Point (RP) when an ACD Queue is associated with that RP. Previously, SIP Server generated EventTreatmentNotApplied for the ACD Queue. (SIP-14652)
SIP Server now correctly parks an agent with a nailed-up connection after a consultation call is released, in the scenario where an agent initiates a consultation call to an external destination using a TInitiateConference request, and after SIP Server sends an INVITE, the agent releases the call by issuing a TReleaseCall request. Previously, in this scenario, the agent connection became stuck. (SIP-14651)
In an Active-Active RM deployment, when both RMs are out-of-service and the
sip-error-conversion
option is configured at the Application
level, if an inbound call arrives to the GVP trunk, SIP Server now correctly
converts and sends the error code to the caller as configured. Previously, in
this scenario, SIP Server did not convert and sent the original error code,
ignoring the option setting. (SIP-14596)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.45. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server can now monitor the health and status of Active Call Recording, and generate an alarm if required.
SIP Server supports a standard log event, 01-52051, for unsuccessful call recording scenarios. A recording scenario is considered unsuccessful if one of the recording operations fails and cannot be recovered by SIP Server. For example, SIP Server tries to start recording on a DN and fails. SIP Server then makes a second attempt to start this recording, using a different MCP. If the second attempt is successful, the alarm is not generated. If the second attempt also fails and recording is not started, then the recording scenario is considered unsuccessful and the alarm is raised.
An alarm can be issued for the start, pause, and resumption of MSML-based recording operations regardless of the method selected to trigger them. For example, an alarm can be triggered for the start recording command if it is submitted from an agent desktop or from the routing strategy in a TRouteCall request, or if it is activated internally based on a DN configuration. An alarm is not raised for the stop recording operation. If an attempt to stop the recording fails, SIP Server terminates recording dialogs without raising an alarm.
This functionality is implemented by using the new
recording-failure-alarm-timeout
option, which is set in the
TServer
section.
recording-failure-alarm-timeout
Section: TServer
Default Value: 0
Valid Values: 0
–65535
(seconds)
Changes Take Effect: Immediately
Enables call recording alarm notification. When this option is set to a value
other than 0
(zero), SIP Server generates a 52051
alarm message and starts the timer using the interval defined by this
configuration option when the first call recording failure is detected. Each
consecutive call recording failure detected during this period increments
the counter.
When the timer expires, SIP Server generates an alarm message with the number of failures detected in the past interval. If the timer expires and no recording failures have been detected within the past interval, SIP Server does not generate an alarm message.
Setting this option to 0
(zero) disables the feature.
Note: This call recording alarm is designed as a persistent alarm. An administrator can clear this alarm manually or use the Clearance Timeout timer in Genesys Administrator.
(SIP-13589)
TLS encryption is now supported between SIP Server and Active-Active Resource Managers in a deployment where SIP Server high-availability is configured using the F5 Networks BIG-IP Local Traffic Manager. (SIP-14059)
This release also includes the following corrections and modifications:
SIP Server now properly sends a call that has been rejected by all members
of the Hunt Group to the default DN, removing that call from the ACD Queue
and enabling new calls to be distributed to the members immediately after
the timeout specified by the hg-busy-timeout
expires.
Previously, SIP Server did not send these rejected calls to the default DN,
and they remained in the ACD queue, blocking all new calls from being
distributed. The calls were removed from the queue only after the original
call was disconnected. (SIP-14739)
SIP Server now correctly plays agent and customer greetings after dynamic recording for the call starts. Previously, greetings sometimes were not started because of a race condition between requests for greeting and dynamic recording. (SIP-14620)
SIP Server now correctly provides the same value in the Expires header and
expires
parameter of the Contact header in 200 OK responses to
REGISTER requests. (SIP-14409)
SIP Server now releases the transferring party in the single-step transfer
scenario when the REFER is Accepted, in the double-dip IMS environment
where the server-role
configuration option is set to 3
and the divert-on-ringing
option is set to false
.
Previously, in this scenario, SIP Server did not release the transferring
party until the transfer destination answered the call. (SIP-14646)
SIP Server no longer returns an EventError "Unknown cause"
message when the Authentication procedure for a REFER request fails. SIP Server will now pass a SIP response code in EventError in accordance with the map-sip-errors option
setting. (SIP-4876, ER# 312768969)
SIP Server no longer starts distributing a new hunt group call until the previous call is answered by a hunt group member and gets diverted from the hunt group. Previously, SIP Server could incorrectly start distributing a new call when the previous one was still not answered. This could lead to a hunt group member having two calls ringing on his/her phone simultaneously. (SIP-16306)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.45. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
cnonce
parameter in
HTTP Digest authentication challenges if it receives the qop
directive in the www-authenticate
header field. Previously,
SIP Server did not include the cnonce
parameter in authentication
challenges. (SIP-14570, ER# 312490411)
This release also includes the following corrections and modifications:
SIP Server now correctly handles the situation in which Hunt Group call
distribution fails if a single member DN does not have the proper SIP
contact, and is therefore not registered. In this case, the call is
readied for distribution to any of those member DNs that are registered;
if no DNs are available within a specified timeout, the call is distributed
to the DN specified by the default-dn
option.
Previously, if one of more DNs in a Hunt Group were not registered with
SIP Server, a call directed to that Hunt Group was stuck; it was not
distributed to any members of the Hunt Group or to the
default-dn
. (SIP-14603)
SIP Server no longer experiences a memory leak when the timeout specified
in extension AttributeTimeout
of TMakePredictiveCall expires
and a call is released at a Routing Point. Previously, in this scenario,
call objects were not released properly resulting in a memory leak.
(SIP-14499)
SIP Server no longer consumes 100% CPU when processing TCP exceptions that occurred if there is only one transport for a trunk and if there are two active SIP transactions for that transport. (SIP-14678)
For Delta-Proxy IMS deployments only: SIP Server now correctly inserts
the orig
parameter in the Route header of INVITE messages
going to a customer and does not insert it in the messages going to an
agent in predictive-call scenarios. Previously, SIP Server was omitting the
orig
parameter in the Route header of INVITE messages going to
both a customer and an agent if the ims-3pcc-prefix
configuration option was defined. (SIP-14512)
SIP Server now correctly reports CPU usage for operational information on MS Windows in the SIP Server Operational Information log. It requires LCA version 8.1.300.13 and later for this log thread to be reported. (SIP-14547)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.44. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server now provides the ability to send AoC (Advice of Charge) notifications only when a call is answered (that is, the destination party is in the established state). It is a regulatory requirement in many countries.
For this feature to work, SIP Server distributes calls through a Routing
Point that is configured with the divert-on-ringing
option
set to false
. A T-Library client that monitors the Routing
Point (for example, URS or ORS) receives the notification that the call
is delivered to the destination when the outgoing call is answered. (SIP
Server sends EventRouteUsed/EventDiverted to its clients). This
notification can be used as a trigger for generating an AoC notification
using a TPrivateService(3018) request.
SIP Server is able to process this request and send a SIP INFO AoC message to the destination even though the Routing Point DN, used to route the call and passed as a value of AttributeThisDN of the TPrivateService(3018) request, is already released from the call.
To implement this functionality, use the following option:
sip-enable-aoc-after-established
Section: TServer
Default Value: false
Valid Values: true
, false
Changes Take Effect: For the next request
When this option is set to true
, it enables the mode of
providing AoC notifications for established calls. In particular, SIP
Server accepts and processes TPrivateService(3018) AoC requests in which
AttributeThisDN refers to a Routing Point DN that is not present in the
call. At the same time, the AOC-Destination-DN
extension key
points to an existing party in the established state. To successfully
process this request, the Routing Point DN referred by AttributeThisDN
must also have the divert-on-ringing
option set to
false
.
When this option is set to false
(for backward compatibility),
SIP Server rejects an AoC TPrivateService(3018) request if AttributeThisDN
refers to a DN that is not present in the call.
(SIP-14134)
In addition to the current TLib-SIP Mapping functionality, SIP Server now supports the EXTRACT_SIP_HEADERS extension key in the TMakeCall, TInitiateTransfer, and TInitiateConference requests. This extension key specifies a comma-separated list of SIP headers. When an INVITE is sent to a TMakeCall/TInitiate originating party, these SIP headers will be mapped to AttributeExtensions of the EventDialing message generated for this party. (SIP-14133)
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly in an MSML configuration when handling TCP socket exceptions that are generated for a transport associated with multiple active transactions. (SIP-14408)
SIP Server now properly maps the AlertInfo
header from a
TRouteCall request to an INVITE request that it sends to members of the Hunt
Group. Previously, SIP Server did not include the AlertInfo
header in these requests. (SIP-14548)
SIP Server now includes the CPNDigits extension key in the INVITE request sent to Hunt Group members after a TRouteCall request (with the CPNDigits extension key) issued by URS. Previously, SIP Server did not add CPNDigits to the INVITE request sent to Hunt Group members; it only added CPNDigits to the INVITE request sent to Media Server. (SIP-14513)
SIP Server now correctly includes the dialed number in the From
header of an INVITE request sent to an agent DN that issues a TMakeCall
request with the CPNDigits extension key. Previously, in this scenario, SIP
Server replaced the destination information in the From
header
with CPNDigits. (SIP-14374)
SIP Server now correctly includes the dialed number in the From
header of an INVITE request sent to an agent DN after an outbound call using
a TMakePredictiveCall request that contains the CPNDigits extension key.
Previously, in this scenario, SIP Server replaced the destination information
in the From
header with CPNDigits. (SIP-14058)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.44. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server enables you to improve the reliability of silence detection in
deployments where CPD is performed by the Genesys Media Server. The
timeguard-reduction
configuration option can be used to make
the time interval used by the Media Server for post-connect CPD shorter
than the time SIP Server is waiting to receive a CPD result from the Media
Server.
Proper configuration can guarantee that SIP Server will always receive CPD
results from the Media Server on time and will never be forced to provide
its own default CPD result when its internal timeout expires. This feature
applies mostly to silence detection because a CPD result of silence takes
the longest time to be detected and the highest risk of SIP Server internal
timer expiration. Use the timeguard-reduction
option to improve
silence detection quality.
timeguard-reduction
Section: TServer
Default Value: 0
Valid Values: 0
–30000
ms
Changes Take Effect: Immediately
Calculates the timer duration that SIP Server sends to Media Server to
limit the time of the post-connect CPD detection. If the original post-connect
CPD timeout value specified by the cpd-info-timeout option
(in
seconds) or call_timeguard_timeout
key in AttributeExtensions
is greater than zero (0
), then the timeout value sent to Media
Server is calculated as follows:
<original CPD post-connect timeout>
- 'timeguard-reduction'
If the calculated value of the post-connect timeout to be sent to the Media
Server is less than 200 ms, then the timeguard-reduction
option
is ignored and the original post-connect CPD timeout value is distributed.
This parameter can be used to improve the reliability of silence detection
in the Outbound Solution. A reduced post-connect CPD timeout in the Media
Server should ensure that the CPD result of silence is received by SIP
Server before its own timer expires. A practical value of the
timeguard-reduction
option can be slightly more than the
round-trip time between SIP Server and Media Server.
An increased value of the timeguard-reduction
option improves
the reliability of silence detection, but at the same time it shortens the
time taken for the CPD post-connect detection for all scenarios. To avoid
this, the value of the original CPD post-connect timeout must also be
increased when timeguard-timeout
is defined. If millisecond
precision is required for the definition of the post-connect CPD timeout
in the Media Server, then the call_timeguard_timeout
key in
AttributeExtensions of TMakePredictiveCall must be used to define the
original post-connect CPD timeout.
AttributeExtensions
Key: call_timeguard_timeout
Value: Integer
Request: TMakePredictiveCall
If set, the value of this Extension specifies the maximum time, in milliseconds, allowed for post-connect CPD results to be received from the device performing CPD (either Genesys Media Server or Media Gateway).
(SIP-6444)
This release includes the following corrections and modifications:
This version of SIP Server is built with TSCP 8.1.010.44, which corrects the following issue:
SIP Server can now control the scope of the requests transmitted to a
Virtual Service Provider (VSP) by setting the new Application-level restricted
vsp-process-udata
option to true
. Previously
(starting in version 8.1.1), SIP Server unexpectedly started to transmit
User Data requests addressed to Virtual Queues to its VSP even though it
had not done so in earlier versions. This new option has been added to
provide backward compatibility.
(SIP-14039)
When both the agent-greeting
and customer-greeting
options are configured for an Agent Login object, and the
greeting-delay-events
option is set to true
, SIP
Server now generates EventEstablished if greetings are not played.
Previously, if greetings were not played because, for example, the Media
Server was out-of-service, SIP Server did not generates EventEstablished
and the call eventually timed out. (SIP-14305)
When MSML-based silent monitoring is in use and the Media Control Platform (MCP) on which the conference was created goes out of service, SIP Server now recovers the monitoring session using an alternate instance of MCP when it receives NOTIFY from Resource Manager about MCP failure. Previously, in this scenario, SIP Server did not recover the session, even when it received NOTIFY. (SIP-14250)
SIP Server now correctly applies continuous treatments when the Media for
a call has changed. Previously, in this scenario, and when the
sip-treatments-continuous
option was set to true
and the divert-on-ringing
was set to false
, SIP
Server did not send the proper notifications, so the caller sometimes heard
silence until the called party answered. (SIP-14264)
In a multi-site supervision scenario, SIP Server now generates
EventMonitoringNextCall with the proper AttributeExtensions for an Agent
DN that is configured in both the local and remote sites and is monitored
by a supervisor in the remote site. Previously, SIP Server removed the
AttributeExtensions and used the default-monitor-mode
and
default-monitor-scope
key values in the AttributeExtensions
for the EventMonitoringNextCall. (SIP-14136)
A nailed-up session is now correctly synchronized between primary and backup
SIP Servers. Previously, after a switchover, SIP Server set up a new nailed-up
connection to a gcti::park
device. (SIP-2632, ER# 285305941)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.42. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates because of call deletion after an unsuccessful TSingleStepConference that was requested by an agent who is involved in two consultation calls with recording. Previously, the call deletion was not handled properly in memory, and SIP Server sometimes terminated unexpectedly. (SIP-14083)
SIP Server now properly handles the scenario in which a Routing Point is disabled or deleted from the Genesys configuration while it is involved in a conference call. Previously, in this scenario, SIP Server became unstable and terminated unexpectedly. (SIP-14248)
In an ASM outbound call scenario, when switchover occurs immediately after Merge Call (Bridging method), SIP Server now correctly processes the re-INVITE request, received from the agent, for putting the call on hold. Previously, in this scenario, SIP Server did not process the request properly. (SIP-13721)
SIP Server performance no longer decreases when the convert-otherdn
option is set to +agentid
. Previously, if SIP Server had a
significant number of configured DNs, setting this option to
+agentid
caused SIP Server performance degradation. (SIP-14262)
SIP Server no longer terminates unexpectedly after receiving a T-Library message that contains a corrupted/malformed TKVList pair in AttributeUserData, and now, in this case, will reply with EventError. (SIP-14187)
SIP Server no longer terminates unexpectedly if the destination SIP Server in an ISCC transaction receives two or more events propagated by ISCC followed by ISCCEventCallDeleted while the transaction is still in progress. Previously, SIP Server sometimes entered into a loop, consuming 100% of the CPU and triggering an Out of Memory error. (SIP-14172)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In an IP Address Takeover high-availability configuration, where a Voice over IP Service DN is configured with the transport TCP and the oos-check
option is set to a small value (that is, Active OOS is enabled), if SIP Server attempts to open a TCP connection before the VIP address is present on the host, it now tries to open the TCP connection again. Previously, in this scenario, SIP Server did not retry to open the TCP connection and considered the DN as out of service. (SIP-14093)
SIP Server no longer sends EventRouteRequest to a Routing Point DN that is disabled in the configuration environment. Previously, it erroneously did. (SIP-13725)
SIP Server no longer terminates abnormally after a switchover from backup to primary mode if a call is diverted from a Routing Point to another DN when the after-routing-timeout
expires. Previously, the call was not properly diverted from the Routing Point, causing SIP Server to terminate after the switchover. (SIP-13815)
SIP Server now correctly generates EventEstablished for a new call to a parked agent with a nailed-up connection when the agent is on hold. Previously, in this scenario, the call was distributed to the agent but SIP Server did not generate EventEstablished. (SIP-14191)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server provides the ability to pass the identity of the party, which has originated the transfer, in the SIP URI of the outgoing REFER request's Referred-By header. In addition, SIP Server provides the ability to control the "hostport" component of the SIP URIs in Refer-To and Referred-By headers of the outgoing REFER requests through the configuration options.
Feature Configuration
Objective | Related Procedures and Actions |
---|---|
1. Configure the SIP Server Application. | Set sip-referred-by-support to true . |
2. (Optional) Configure a DN associated with the transferred/routed party. |
If it is required to override the "hostport" component of the SIP URI in Refer-To and Referred-By headers, configure the following options on a DN where an outgoing REFER request is sent to:
|
Override of the "hostport" component of the SIP URI in the Refer-To header can be configured by the following options on a DN where an outgoing REFER request is sent to, in order of priority:
override-domain-oosp
, in case of OOSP transfer override-domain-refer-to
override-domain
Override of the "hostport" component of the SIP URI in the Referred-By header can be configured by the following options on a DN where an outgoing REFER request is sent to, in order of priority:
override-domain-referred-by
override-domain
Configuration Options
To support this feature, the following options were introduced:
sip-referred-by-support
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Application-level option. If set to true
, SIP Server sends an identity of the party, which has originated the transfer, in the SIP URI of the outgoing REFER request's Referred-By header. If the call is processed on the Routing Point and is routed using TRouteCall, then the Routing Point name is used as a userpart of the Referred-By SIP URI. In addition, SIP Server can substitute the "hostport" component of the SIP URI in Refer-To and Referred-By headers of REFER messages with the values configured by a user. If set to false
, this feature is disabled.
override-domain-refer-to
Section: TServer
Default Value: NULL
Valid Values: Any computer name string
Changes Take Effect: For the next call
DN-level option that must be configured on a DN associated with the transferred/routed party where REFER is sent to. If set, SIP Server substitutes the "hostport" component of the SIP URI passed in the Refer-To header of the outgoing REFER request with the value of this option. Applies only if sip-referred-by-support
is set to true
.
override-domain-referred-by
Section: TServer
Default Value: NULL
Valid Values: Any computer name string
Changes Take Effect: For the next call
DN-level option that must be configured on a DN associated with the transferred/routed party where REFER is sent to. If set, SIP Server substitutes the "hostport" component of the SIP URI passed in the Referred-By header of the outgoing REFER request with the value of this option. Applies only if sip-referred-by-support
is set to true
.
(SIP-13575)
This release includes the following corrections and modifications:
When SIP Server sends a message longer than 1024 bytes to an unreachable host that is in the same IP network sub-net, and no static ARP is configured, there will be no delay in sending the next message to the reachable destination.
In this case, you must configure the following registry on the Windows OS:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\services\AFD\Parameters\FastSendDatagramThreshold = (DWORD) 4096
(SIP-2814)
If a two-step transfer is completed successfully and SIP Server receives a TReleaseCall request containing the Connection ID of the consultation call, SIP Server now rejects the request with the Event Error (Invalid Connection ID) message. Previously, SIP Server incorrectly released the main call. (SIP-6636)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In multi-site supervision scenarios with MSML configuration, where a supervisor at the remote site monitors an agent located at a local site, SIP Server now correctly sets the gvp:confrole
for the supervisor in the INFO message: monitor
(for Mute mode) or coach
(for Coach mode). Previously, SIP Server did not update the gvp:confrole
for the supervisor. (SIP-13724)
In the SIP Server with Active-Active RM pair deployment, SIP Server now correctly applies the oos-check
when sending the OPTIONS messages to the RM pair, in cases where an MSML Voice over IP service DN is configured with the contact-list
option. Previously, SIP Server did not send OPTIONS messages to the second RM in the list, if the first RM was in the active call. (SIP-13742)
In some scenarios with the softswitch configuration, where oos-force
and oos-check
configuration options are set to valid values, SIP Server now correctly uses the new contact information specified in the 18x early media reliable response when sending the PRACK message. Previously, SIP Server used the original contact information to send the PRACK. (SIP-13764)
If an agent with a nailed-up connection is involved in main and consultation calls that are placed on hold, and the 2nd party releases the consultation call when there is an attempt to retrieve it, SIP Server now correctly releases the consultation call and places the agent back on the main call, which remains on hold. Previously, in this scenario, the agent was not re-invited to the main call. (SIP-6610)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
If the value of the logout-on-disconnect
option is set to false
, SIP Server no longer terminates when an agent DN with the logged-in agent is deleted from the configuration environment. (SIP-13827)
SIP Server now responds with EventAgentLogin to consecutive TAgentLogin requests from T-Library clients, if those TAgentLogin requests contain values for the AgentID, ThisQueue, and ThisDN attributes matching those in the first TAgentLogin request. EventAgentReady and EventAgentNotReady are populated with the current agent state. Previously, SIP Server distributed EventError in response to consecutive TAgentLogin requests. (SIP-6083)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In multi-site scenarios, SIP Server that is located in a signaling path between other SIP Servers can now pass 183 Session in Progress
messages with Advice of Charge (AoC) information. (SIP-6019)
Starting with version 8.1.100.74, SIP Server supports sending the Advice of Charge (AoC) information using 183 Session in Progress
reliable provisional responses for non-established dialogs of inbound calls. This feature is enabled by setting the sip-enable-100rel
option to true
either at an Application level or at a DN level where setting at the DN level takes precedence. Previously, SIP Server did not enable the feature if the
DN-level sip-enable-100rel
option was set to true
, but the Application-level sip-enable-100rel
option was set to
false
. (SIP-6049, ER# 321260142)
SIP Server now correctly removes the capacity value from its memory when the capacity
configuration option is removed from the Trunk on which it was set. Previously, SIP Server did not remove the capacity value from its memory though it was removed from the Trunk. (SIP-13626)
In a multi-site scenario using the direct-uui
transaction type, SIP Server now correctly passes the X-ISCC-Id header received in a REFER request to a corresponding INVITE request, if the sip-pass-refer-headers
option is set to X-ISCC-Id
. Previously, SIP Server did not pass the X-ISCC-Id header, which resulted in a mismatched call. (SIP-13810)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature:
Log expiration. Automatic log deletion upon expiration is controlled by the expire
option. One of the ways to control log expiration is to set a maximum number of files to store. In this case, the limit is applied to each thread separately. For example, if sip-link-type=3
, multi-threaded logging is enabled and the limit is set to 30
, SIP Server keeps approximately 90 files, at least; the rest are considered expired. The three threads (T-Server, Call Manager, and Transport), each contribute 30 files toward the total. (SIP-13572)
This release includes the following corrections and modifications:
After a switchover, SIP Server now correctly uses the Expires
header provided in a REGISTER request. Previously, SIP Server did not apply the Expires
header content after its switchover. (SIP-6649)
SIP Server now correctly processes incoming concurrent re-INVITE requests when the session refresh is in progress. Previously, SIP Server sometimes did not process the concurrent re-INVITEs correctly, which resulted in a dropped call. (SIP-6277)
SIP Server no longer drops an original call and correctly processes 4xx, 5xx, or 6xx SIP responses from an external destination resulting from a single-step transfer to that destination made through a Trunk configured with reuse-sdp-on-reinvite=true
. Previously, in this scenario, SIP Server dropped the original call after the unsuccessful transfer. (SIP-5969)
In outbound-call scenarios initiated by an agent with early media established, SIP Server now correctly handles race conditions involving 2xx successful responses between parties when the agent places the call on hold and then retrieves it. Previously, in this scenario, SIP Server dropped the call. (SIP-5743, SIP-6411)
SIP Server is now able to apply an MSML-based treatment (greeting) to a call when NETANN-based recording of the established call fails because of the recorder being out-of-service. Previously, in this scenario, SIP did not start other available services. (SIP-5688)
If SIP Server receives REFER with Replaces while establishing a call treatment with Media Server and a switchover occurs, the call is now correctly synchronized after the switchover. (SIP-5143)
SIP Server now correctly maps User Data containing multi-line characters (including carriage returns and line breaks) to an RFC-compliant SIP message when the userdata-map-format
option is set to sip-headers-encoded
on the Voice over IP Service DN. Previously, SIP Server did not map those characters correctly. (SIP-6401)
In a multi-site configuration with the route
ISCC transaction type and the divert-on-ringing
option set to false
at an Application level, when a call arrives through an External Routing Point and is routed to an agent, and then the call is redirected to an internal Routing Point because of the no-answer timeout, SIP Server now correctly reports EventRouteUsed and EventDiverted. Previously, SIP Server did not report those events. (SIP-13645)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly processes geo-location information in certain msml scenarios. Previously, when SIP Server received a TApplyTreatment request that did not include geo-location in the AttributeExtensions, followed by a second TApplyTreatment request that did include the geo-location, SIP Server used only the existing (first) connection to process the treatment, making the geo-location from the second request unavailable for resource selection. Now, based on the second request, SIP Server establishes a new connection with the Resource Manager, adding the geo-location info in the X-Genesys-geo-location
header, so that SIP Server is able to use geo-location when selecting Resource Manager and MCP. (SIP-6640, ER# 324893166)
SIP Server is now able to successfully recover music-on-hold (MOH) service after a high-availability (HA) switchover when it is working with an active-active Resource Manager pair configured in the contact-list mode. Previously, when a switchover occurred while MOH was playing, and the MCP used for MOH became unavailable (Resource Manager notifying SIP Server of the outage), SIP Server was unable to recover the service. Now, on getting notification from Resource Manager that the MCP is down in this scenario, SIP Server is able to successfully recover the service. (SIP-6524, ER# 323837272)
SIP Server now correctly handles call path optimization scenarios. Previously, SIP Server could experience performance degradation and memory growth in complex Inter Service Call Control (ISCC) call scenarios that involved call path optimization (out-of-service-path transfers) and ISCC loop detection (if the value of the match-call-once
option was set to false
). (SIP-6641, ER# 324893117)
SIP Server now correctly sends subsequent INFO messages when applying a second treatment to an available Resource Manager (RM). Previously, in active-active Resource Manager scenarios (deployed in contact-list mode), if an RM instance failed after the first treatment was applied to a call on a Routing Point, SIP Server sometimes tried to send INFO messages to the failed RM instance (even though the RM was detected as Out of Service (OOS)), causing a subsequent second treatment to fail. Now, when processing the second treatment, SIP Server sends the subsequent INFO message to the available RM instance. (SIP-6529, ER# 323895621)
SIP Server no longer incorrectly attempts to resolve the special contact ::msml
setting, in cases where oos-check
and oos-force
options are configured on the Trunk DN. Previously, when OOS detection was enabled, SIP Server inadvertently tried to resolve the DNS name msml
, resulting in failure.(SIP-6414, ER# 322798651)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.38. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new functionality:
Filter greetings by call type. SIP Server now lets you suppress agent greetings for different call types. You can block greetings for internal, consultation, and outbound calls, either globally at the Application-level, or individually per Agent Login.
To configure this feature, configure the following options:
true
in cases where SIP Server routes calls via ISCC to another SIP Server instance.
none
, which plays greetings to the agent regardless of call type (overriding any restrictions set globally at the Application-level).
To support this feature, the new values has been added to the existing configuration option greeting-call-type-filter
. In addition, the option can now be set at the Agent-Login level, as follows:
Application-level
greeting-call-type-filter
Section: TServer
Default Value: No default value
Valid Values: internal, consult, outbound
Changes Take Effect: Immediately
Specifies the types of calls for which greetings will not be played. You can set this option to multiple call types, using a space, comma, or semicolon-separated list. By default (option has no value), greetings are played for all calls (this provides backwards compatibility to SIP Server release 8.0.2). You can suppress greetings for internal, outbound, and consultation calls, using these values: internal
, outbound
, and consult
.
Agent-Login level
greeting-call-type-filter
Section: TServer
Default Value: No default value
Valid Values: none, internal, consult, outbound
Changes Take Effect: Immediately
Specifies the types of calls for which greetings will not be played. You can set this option to multiple call types, using a space, comma, or semicolon-separated list. The agent login setting takes precedence over the Application-level setting. You can suppress greetings for internal, outbound, and consultation calls to this agent, using these values: internal
, outbound
, and consult
. If you set the option to none
, the greeting is played to the agent regardless of call type. This keyword none
cannot be used in conjunction with other values or delimiters. If this option is incorrectly configured on the Agent-Login level, SIP Server disregards this option setting, using the Application-level option setting instead (if it is configured).
This release includes no following corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.38. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new functionality:
Check for updates
button if
a newer version of SIP Server is available. When you click the button, the
SIP Server Installation Package window displays, containing links to the
Release Notes to view the updates for newer available 8.1 versions of SIP
Server.This release also contains the following correction or modification.
SIP Server no longer terminates unexpectedly in a multi-site scenario in
which a call is routed from Site A to Site B and back to Site A, if the
oosp-transfer-enabled
or sip-server-inter-trunk
option is set to true
for the Trunks pointing to the peer
SIP Servers. Previously, in this scenario, SIP Server at Site A terminated
unexpectedly when it received the 302 message for the second routing.
(ER# 324789161)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.37. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In early media scenarios, SIP Server no longer starts call monitoring until a call is established between a caller and an agent. Previously, SIP Server started call monitoring immediately after Hold and Retrieve operations and before the call was established. (ER# 323631867)
SIP Server no longer drops the call in the following scenario:
X-Genesys-CallInfo: routed
header.In blind-transfer scenarios in which an early media dialog is established with a transfer destination, and if the transfer destination rejects the call with a 603 Decline message during the transfer completion, SIP Server now correctly generates proper events after the transferring party has left the call. Previously, in this scenario, SIP Server did not generate an EventAbandoned message for the transfer destination. (ER# 319948398)
SIP Server no longer terminates when receiving SIP messages without the Contact header parameters in response to the INVITE with the transport=udp
parameter in the Request-URI. Previously, in this scenario, SIP Server sometimes terminated. (ER# 321586860)
If, while in a conference, an agent issues TReleaseCall and TDeleteFromConference requests simultaneously, SIP Server now releases the agent from the call. Previously, in this scenario, when running in the multi-threaded mode, SIP Server terminated unexpectedly. (ER# 323294813)
In predictive-call scenarios, when Call Progress Analysis (CPA) is in progress and SIP Server receives a BYE message from the outbound Trunk, SIP Server now generates an EventReleased message with the CallState attribute set to 5
(remote release) and GSW_CONNECT_TIME in the AttributeExtension. Previously, SIP Server generated EventReleased with CallState set to 0
. (ER# 322871385)
For Delta-Proxy IMS deployments only: SIP Server no longer sets the orig
parameter in the Route header of INVITE messages going to an agent in predictive-call scenarios. Previously, in these scenarios, the orig
parameter was set in the INVITE messages going to both a customer and an agent. (ER# 322871393)
In the SIP Server with Active-Active RM pair deployment using the IP-based model (using the contact-list
option), when a conference is established and switchover occurs, SIP Server is now able to add or modify parties in the conference. Previously, SIP Server was not able to add or modify parties in the conference. (ER# 321932663)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.36. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in this release of SIP Server.
When the Voice over IP Service DN with service-type=voicemail
is configured with contact=::msml
, SIP Server can retrieve
a contact from the msml VOIP Service DN and select an active RM to which
to send the voicemail request. (ER# 321998591)
SIP Server now accepts the Message Waiting Indicator (MWI) subscription
for a VoiceMailBox if the new Application-level option
mwi-subscribe-vmb
is set to true
. If this option
is not present, or is set to false
, SIP Server only accepts
MWI subscriptions for DNs.
mwi-subscribe-vmb
Section: TServer
Default Value: false
Valid Values: false
, true
Changes Take Effect: Immediately
Specifies whether SIP Server accepts Message Waiting Indicator (MWI)
subscriptions that are submitted for a VoiceMailBox (VMB). If set to
true
, SIP Server accepts MWI subscriptions for VMB numbers
and for DNs. If set to false
(the default), SIP Server
accepts MWI subscriptions for only DNs.
In addition, the mwi-implicit-notify
option must be set to
false
, and the mwi-notify-unregistered-dn
option
must be set to true
to support this feature.
Note: This feature requires SIP Feature Server version 8.1.2 or later.
(ER# 321087863)
Support of SIP Voicemail for SIP Feature Server in a Business Continuity deployment. This feature requires SIP Feature Server version 8.1.2 or later. SIP Server also supports the Dial Plan feature implemented in SIP Feature Server. That way, the dial plan can be configured either on the SIP Server side (as before) or on the SIP Feature Server side. Refer to the SIP Feature Server 8.1.2 documentation for details.
SIP Server supports the following Voicemail features:
Configuration: Create a Voice over IP Service DN with the following options:
service-type
—Set to voicemail
.contact
—Set to the FQDN of Resource Manager.(ER# 317155848)
Support of Inbound voice via CUBE SBC version 15.2(4)S0b, release software (fc1). SIP endpoint's SIP registration via CUBE SBC (registration pass-through) is not currently supported.
SIP Server supports the new configuration option: sip-error-overflow
.
TServer
DN-level option. Specifies the destination number to which SIP Server will forward a call if this device responds with a failure (error) to a SIP INVITE.
Notes:sip-error-overflow
option is defined and an error response is received from the external destination, the outbound call is transferred to the overflow DN. sip-error-overflow
configuration option. sip-busy-type
configuration option is not applicable, if sip-error-overflow
is configured, since the device can no longer be busy (it will forward to sip-error-overflow
destination instead).sip-error-overflow=gcti:voicemail
is not supported.This release also includes the following corrections and modifications:
SIP Server no longer grows in memory in the ISCC Call Overflow scenarios.
Previously, when a call was routed by ISCC using the route
transaction type and the Call Overflow feature was activated, SIP Server
grew in memory. (ER# 322871400)
When running on the Linux, Solaris, or AIX platform, SIP Server no longer terminates unexpectedly if a TRequest is rejected because of overload conditions. (ER# 322708670)
SIP Server now successfully routes a call to default-dn if an external destination is configured as default-dn and the call is routed to the default-dn immediately after arriving on a Routing Point because of URS unavailability. Previously, in this scenario, SIP Server released the call. (ER# 323036948)
SIP Server no longer terminates unexpectedly while parsing the
userdata
list in the header of an INFO message when the last
key-value pair in the list contains only a key but no value after the "="
sign. Previously, in this scenario, SIP Server terminated unexpectedly.
(ER# 322461234)
SIP Server can now generate a SUBSCRIBE request if the option contact on the softswitch Voice over IP Service DN does not contain a port number. Previously, in this scenario, SIP Server was unable to send the SUBSCRIBE request. (ER# 322383104)
Now, only one configuration section called log
is created in
the SIP Server application when the application is created using Genesys
Administrator and the SIP Server XML template file. Previously, in this
scenario, duplicate log sections were created, one called log
and one called Log
. (ER# 322174205)
SIP Server no longer sends empty MWI (Message Waiting Indicator) notifications within NOTIFY messages to SIP clients, and inside EventUserEvent messages to T-Library clients, when an agent logs in or logs out. (ER# 321583241)
SIP Server is now able to retry a subscription on another Media Server
in 30 seconds after SIP Server considers the TCP transport as failed.
This issue applied to a single-threaded SIP Server configured with
sip-link-type=0
(zero). Previously, SIP Server was unable
to retry a subscription on another Media Server. (ER# 321530580)
SIP Server in high-availability mode, when integrated with Cisco Unified Communications Manager 7.1, now correctly synchronizes the agent state after a switchover occurs. Previously, if an agent was set to the Logout or NotReady state, after a switchover SIP Server erroneously set the agent back to the Login/Ready state. (ER# 321459088)
In addition to the current supported Advice of Charge (AoC) feature where
SIP Server uses the INFO method to send the AoC information, SIP Server can
now send the AoC information using 183 Session in Progress
reliable provisional responses for non-established dialogs of inbound calls.
To enable this feature, the sip-enable-100rel
option must be
set to true
at the origination device (Trunk) or at the SIP
Server Application level. Setting this option on the device (Trunk), if
configured explicitly, takes precedence over the Application-level option.
SIP Server receives the AoC information in a TPrivateService
message sent from a T-Library client for a non-established dialog. SIP
Server maps the AoC information from TPrivateService
to a
183
message instead of the INFO message. If AoC is successfully
delivered, then SIP Server responds with EventACK. Otherwise, it responds
with EventError.
Note: SIP Server does not enable the feature if the
DN-level sip-enable-100rel
option is set to true
, but the Application-level sip-enable-100rel
option is set to
false
. This issue was fixed in version 8.1.100.82.
(ER# 321260142)
SIP Server now supports the BroadWorks SIP Trunking solution by using the Network Asserted Identity mechanism for controlling the presentation of personal information (caller details) in SIP messages within a trusted network. The following SIP headers are included when SIP Server generates outgoing SIP INVITE messages:
P-Asserted-Identity
header must contain the Address of
Record (AOR) of the trunk. The header must be present regardless of the
privacy settings. The value must be explicitly configured on a Trunk level.From
header must contain a valid identity of a caller
known to both the Carrier's SIP Network and SIP Server, even when CLIR
(anonymous call) is requested. This identity could be a specific user, or
a number/identity associated with an outbound campaign or a routing point.Privacy
header must be present,
containing the option-specified value (user,id
if the
recommended configuration is used).To configure this feature, set the following configuration options:
enable-preserve-privacy=true
network-provided-privacy=user,id
enforce-p-asserted-identity=<
Address of Record (AOR) of the
trunk>
enforce-trusted=true
enable-preserve-privacy
Section: TServer
Default Value: false
Valid Values: true
, false
Changes Take Effect: At the next call
When set to true
, the privacy service requested by an initial
call is preserved during routing, transfer, consultation, and conference
operations. When set to false
, those operations reset the call
privacy state, unless it is explicitly requested by the origination DN.
network-provided-privacy
Section: TServer
Default Value: An empty string
Valid Values: A comma-separated list of privacy type tags, such as
user,id
Changes Take Effect: At the next call
DN-level option for Trunk
DNs. If configured, all outgoing
INVITE messages sent through the trunk are altered in the following way:
From
header of an incoming INVITE request contains
the valid caller ID information, it will not be replaced with an anonymous
tag, even if privacy is requested.From
header contains an anonymous content and the
P-Asserted-Identity
header is present, SIP Server replaces the
anonymous value of the From
header with the
P-Asserted-Identity
header content.From
header contains an anonymous content and the
P-Asserted-Identity
header is not present, SIP Server
passes the From
header with the anonymous content.From
header contains the caller DN information, it is altered if the CPNDigits
override is requested.Privacy
header, if requested, will be
overridden with the option value.P-Asserted-Identity
header will contain the
value of the enforce-p-asserted-identity
option, if it is
configured on the Trunk DN.Feature Limitations:
P-Preferred-Identity
header of the incoming INVITE
message will not be taken into account to resolve the caller ID
information.Privacy
header of the incoming INVITE
and corresponding configuration options privacy
and
enforce-privacy
, which SIP Server accepts, is id
.
Any other value will not be interpreted as a privacy request.override-domain-from
option has no effect on an outbound
Trunk, if a call is initiated by an incoming anonymous INVITE with the
P-Asserted-Identity
header containing a real caller ID.
(ER# 321020987)privacy
option will not be taken into account for routed calls. (ER# 320382287)P-Asserted-Identity
header will always contain the value of the destination trunk option
enforce-p-asserted-identity
and the origination DN option
p-asserted-identity
has no effect on the header value.(ER# 321260135)
The backup SIP Server no longer terminates during the creation of the
16,777,215th call if the sip-link-type
option is set to
0
(zero). (ER# 321259958)
Outgoing in-dialog requests via inter-site connections
(sip-server-inter-trunk
must be set to true
for
the Trunk DN to the other SIP Server) are now routed to an active SIP Proxy
that belongs to the local pool of SIP Proxies. Previously, the requests were
routed to the SIP Proxy instance that was a proxy for the SIP Server that
initiated the dialog. (ER# 321102491)
SIP Server now processes a SIP or T-Library NIC failure properly. By converting FQDNs to the correct IPv4 or IPv6 addresses, SIP Server can now query SIP NIC or T-Library NIC status correctly and respond appropriately. Previously, SIP Server was unable to process either failure in an IPv6 network. (ER# 321064951)
SIP Server now uses the updated value of the contact-list
option of the msml VOIP Service DN and sends OPTIONS messages to the new
IP address. Previously, SIP Server would not recognize the updated value,
and continue to send OPTIONS messages to the previous IP address.
(ER# 321060371)
For deployments in which SIP Server is integrated with SIP Proxy, SIP Server now accepts SIP requests from non-local pools of SIP Proxies. Previously, SIP requests from non-local pools of SIP Proxies were rejected. (ER# 321009721)
If the value of the logout-on-out-of-service
configuration
option is set to true
, SIP Server now logs out an agent if the
agent's DN is deleted from the configuration environment. Previously, in this
scenario, SIP Server became unstable. (ER# 320915874)
SIP Server now drops a call queued on a Routing Point immediately after the Routing Point is deleted from the configuration environment. Previously, SIP Server did not drop the call. (ER# 320864957)
SIP Server now drops a call queued on an ACD Queue immediately after the ACD Queue is deleted from the configuration environment. Previously, SIP Server did not drop the call. (ER# 320864800)
When SIP Server is integrated with Media Server in Active-Active RM IP mode
(using the contact-list
option) and contact of the Trunk object
is configured as ::msml
, the request URI of the INVITE sent to
the Trunk now correctly contains the userpart received in the incoming INVITE.
Previously, it contained the value of the msml VOIP Service DN.
In addition, if using TCP transport when the OPTIONS to one RM fails, that RM is now re-tried when it comes back into service. Previously, that RM was not re-tried.
(ER# 320082880)
SIP Server now correctly deletes old log files according to the value
set in the expire
option setting. Previously, logs were not
removed when the value of the expire option was set to
<number>
of days. (ER# 319982219)
SIP Server now correctly sets AttributeReferenceID in the EventRouteUsed
message in the following multi-site scenario (the route
ISCC
transaction type):
Previously, in this scenario, the EventRouteUsed message was generated, but did not contain AttributeReferenceID.
(ER# 319982212)
In case of Active-Active RM deployment, SIP Server now sends requests to GVP using the TCP transport configured in the Trunk Group. Previously, SIP Server sent requests using UDP transport even if the Trunk Group was configured with TCP. (ER# 319329841)
SIP Server now support the values 0
and max
for
the num-of-licenses
and num-sdn-licenses
configuration options. Previously, you had to explicitly specify the number
of licenses of a certain type allocated for a particular SIP Server.
(ER# 319120151)
On the Linux platform, when a connection to a remote Local Control Agent (LCA) is not established, the connection on LINUX is now closed properly. Previously, the connection was not closed properly, causing SIP Server to consume 100% of the CPU.(ER# 318986320)
When SIP Server is integrated with Media Server in Active-Active RM IP
mode (using the contact-list
option), SIP Server no longer
updates the request-uri
field of the mid-dialog request when
sending the request using alternate transport. Previously, in this scenario,
SIP Server updated the request-uri
field with the contact of
the new transport being used. This resulted in the request being rejected
by the RM with the message 482 Loop detected
.
(ER# 318906020)
SIP Server now includes a correct contact header in the implicit MWI Notify
message sent to the phone. Previously, the message contained an incorrect
header, and a 400 Bad request
error was generated on the phone.
(ER# 318616855)
SIP Server now considers the recording failure INFO message received from a Media Server and retries recording with another Media Server in the following scenario:
Previously, in this scenario, SIP Server ignored the recording failure INFO message.
(ER# 317759174)
SIP Server is now able to deliver voicemail to the logged-in agent to the voicemail box number configured in the Extension object. SIP Server now attaches a VoiceMailbox number in the INVITE message (as the userpart of the To header) that is sent to RM. Previously, SIP Server did not use a correct VoiceMailbox number, and failed to deliver the voicemail. (ER# 317672571)
If the dial plan fails and the dial plan is set to reject
,
SIP Server now plays a busy tone and releases the call. Previously, for
backward compatibility purposes, if the dial plan was set to
reject
, the sip-busy-type
option had to be set
to 2
for SIP Server to reject a call. This requirement no
longer applies. (ER# 317237570)
SIP Server is now able to add a remote supervisor to the call. This issue occurred only if SIP Server was integrated with Media Server working by NETANN. Previously, SIP Server failed to add a remote supervisor to a call. (ER# 317204063)
Now, in a HA configuration, with sip-iptakeover-monitoring
set to true
, if the backup SIP Server detects that the IP
address specified by the sip-address
option is a duplicate of
a Virtual IP address present on the backup host, the Virtual IP address down control
script is executed to remove that IP address on the backup host. In that
case, the Virtual IP address up control script on the primary host is executed.
Note: On the Solaris platform, if you experience duplicate Virtual IP addresses, some remote hosts may be unable to communicate with SIP Server using that Virtual IP address.
(ER# 316666221)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.010.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in the 8.1.1 release of SIP Server.
Overload control mechanism support. To prevent SIP Server encountering a situation with an excessive number of incoming T-Requests, the T-Requests rate is limited by a capacity threshold that is configured by using the following options:
If SIP Server encounters an overload situation with an excessive number of incoming T-Requests, it first sends a warning that states that the capacity level has been reached. If the load continues to increase, SIP Server will take a graceful action to reduce the load. If a T-Requests rate exceeds the capacity, SIP Server rejects all excessive T-Requests with EventError
(error code TERR_SERV_UNAVAIL118
and the following text: Request rate exceeded threshold
).
If the T-Requests rate for a given call exceeds capacity, SIP Server rejects either the corresponding or all excessive T-Requests for the given overloaded calls by sending the following: EventError
(error code TERR_SERV_UNAVAIL118
and text Request rate exceeded threshold
).
If a particular (UserData or ApplyTreatment) T-Requests rate for a given call exceeds capacity, SIP Server rejects excessive corresponding T-Requests for the given overloaded call with EventError
(error code TERR_SERV_UNAVAIL118
and text Request rate exceeded threshold
).
SIP Server continues processing T-Requests for all other calls. It sends a warning canceled message once the T-Requests volume is reduced to an acceptable level for non-call-related thresholds.
overload-ctrl-trequests-rate
Section: TServer
Default Value: 1000
Valid Values: 0-10000
Changes Take Effect: Immediately
Related Feature: ''Overload Control''
Specifies the T-Request rate (T-Requests/sec) that SIP Server is able to maintain without performance degradation. For example, if SIP Server is deployed on host machines with a slow CPU, then set this option to a lesser value. If the T-Request rate exceeds a configured value, SIP Server first sends a warning and, if T-Requests continue to increase, SIP Server rejects excessive T-Requests. Setting the value of the option to 0
(zero) disables this functionality.
overload-ctrl-call-trequests-rate
Section: TServer
Default Value: 10
Valid Values: 1-1000
Changes Take Effect: Immediately
Related Feature: ''Overload Control''
Specifies the T-Request rate (T-Requests/sec) that is allowed for each call. This prevents performance degradation if particular clients issue too many requests. If the T-Request rate for any particular call exceeds the configured value, SIP Server first sends a warning and, if T-Requests continues to increase, SIP Server rejects excessive T-Requests for that call. Setting the value of the option to 0
(zero) disables this functionality/
overload-ctrl-call-tupdateuserdata-requests-rate
Section: TServer
Default Value: 10
Valid Values: 1-1000
Changes Take Effect: Immediately
Related Feature: ''Overload Control''
Specifies the User Data update T-Request (TAttachUserData, TUpdateUserData, TDeleteUserData, TDeletePair
) rate (T-Requests/sec) that is allowed for each call. If the T-Request rate for any particular call exceeds the configured value, SIP Server first sends a warning and, if T-Requests continues to increase, SIP Server rejects excessive user data T-Requests for a particular call. Setting the value of the option to 0
(zero) disables this functionality.
overload-ctrl-call-tapplytreatment-requests-rate
Section: TServer
Default Value: 10
Valid Values: 1-1000
Changes Take Effect: Immediately
Related Feature: ''Overload Control''
Specifies the TApplyTreatment
request rate (T-Requests/sec) that is allowed for each call. If the T-Request rate for any particular call exceeds the configured value, SIP Server first sends a warning and, if T-Requests continues to increase, SIP Server rejects excessive TApplyTreatment
requests for that particular call. Setting the value of the option to 0
(zero) disables this functionality.
(ER# 283602575)
In multi-site call-monitoring scenarios, SIP Server now reports the DN of a call originator in the ANI
and OtherDN
attributes of the EventRinging
and EventEstablished
messages for the supervisor DN. Previously, in these scenarios, SIP Server reported ConfID
in the same attributes of the these messages. (ER# 295688204)
SIP Server now adds more detailed information to the Exception on listen socket
message when writing it to the log. (ER# 298812460)
The new override-from-on-conf
configuration option controls the value of the username in the From
header for outgoing INVITE messages to a new party DN that is added in the single-step conference. If this option is set to false
(the default), the username is set to conf=conf-id
or msml=conf-id
. If this option is set to true
, SIP Server takes the value of the conference initiator DN as the username part of the From
header. (ER# 315202888)
From
header in the initial INVITE request. To enable the feature:
cof-feature
to true
in the extrouter
section.match-ani, match-flexible=false
.When ISCC/COF ANI matching is used, the Number Translation feature in SIP Server can provide more flexibility for handling calls across multiple sites.
(SIP-14680)This release includes the following corrections and modifications:
In No-Answer-Supervision scenarios where the NO_ANSWER_ACTION
parameter is set to notready
, SIP Server now correctly applies the agent status and ignores the Agent Presence notification until the agent takes an action. (ER# 261657915)
The new Application-level option, call-monitor-acw
, enables the application of emulated After Call Work status to a service observer (supervisor).
call-monitor-acw
Default: false
Valid Values: true, false
Changes Take Effect: For the next call
When set to true
, SIP Server applies emulated After Call Work (ACW) to a service observer (supervisor) after a call is released.
(ER# 262519207)
SIP Server now correctly processes a TPlayAnnouncementAndDigits
request with a list of interruptible announcements, and controls playing the second prompt in the list by using Media Server (MSML-based services), if the first prompt is completed and no DTMF digit is received. (ER# 273090851)
SIP Server now correctly handles the following scenario where a call is placed on hold by a THoldCall
request and refresh re-INVITE messages with a hold SDP arrives. Previously, if the value of the sip-cti-control
option was set to talk
, SIP Server misinterpreted refresh re-INVITE messages with the hold SDP as 1pcc hold operation messages. (ER# 274818145)
SIP Server now correctly releases the transferring party before sending a REFER message when performing a single-step transfer by using the pullback
transaction type. Previously, if the value of the refer-enabled
option was set to true
, SIP Server released the transferring party after the REFER message was sent. (ER# 275904559)
SIP Server no longer disconnects a call if it is placed on hold during an agent or customer greeting. Now, when the call is placed on hold during the greeting, the greeting stream is disconnected, and the call remains on hold. (ER# 276010650)
SIP Server no longer plays a ringback tone if a treatment is already applied to the call. Previously, SIP Server incorrectly played a busy treatment and a ringback tone to the transferring party simultaneously. (ER# 276662249)
In a scenario where a Routing Point is associated with an ACD Queue, a call is routed to a busy destination DN (a 486 Busy Here
message is received), SIP Server now routes the call back to the Routing Point, so the strategy on the Routing Point performs the next action for the call. Previously, in this scenario, SIP Server incorrectly routed a call to both the Routing Point and the ACD Queue, which resulted in two strategies being applied to the same call. (ER# 277219813)
If a Voice over IP Service
DN that is configured with the value of the use-register-for-service-state
option set to true
, and it is placed out-of-service, SIP Server now keeps the DN in this state until a REGISTER request for it arrives. (ER# 278526987)
If a hold re-INVITE request is issued by a media gateway, to which SIP Server replies with a 200 OK message, and if SIP Server receives a REFER request before it sends an ACK message in response to the 200 OK, SIP Server now postpones processing the REFER request until the ACK message is received. Previously, in this scenario, if SIP Server received the REFER request before the ACK message, SIP Server dropped the call. (ER# 278579261)
SIP Server is now able to perform SIP-to-TLib mapping properly when the value of the mapped SIP header contains a comma. Previously, SIP Server mapped only the characters that followed the comma into the TLib UserData KVP. Now SIP Server is able to map the entire SIP header value. To activate this functionality, a device to or from which the SIP message is sent or received must be configured with the userdata-map-format
option set to the value of sip-headers-encoded
. (ER# 282373421)
SIP Server now continuously plays a URS busy treatment (in continuous mode) until the routing destination is answered the call. Previously, in this scenario, SIP Server started to play a ringback tone when it received a 180 Ringing
message from a destination DN. (ER# 284258386)
If the make-call-rfc3725-flow=2
and sip-ring-tone-mode=1
options are configured for a caller's DN, SIP Server now correctly plays a ringback tone to that caller. (ER# 285219974)
If the value of the predictive-call-router-timeout
option is explicitly set to the default value of 0
(zero), SIP Server no longer incorrectly prints the Configuration option has invalid value
error message in the log. (ER# 285315841)
While processing a TApplyTreatment
request, SIP Server now correctly processes a 491 Request Pending
from Media Server in response to an INVITE request and also correctly responds to a TRouteCall
request from URS for the same call. Previously, SIP Server responded to TRouteCall
with EventError
. (ER# 285697658)
SIP Server now properly generates an EventPartyDeleted
message, if a party, which was created by substituting the existing party while processing the INVITE with Replaces, is released. Previously, in this scenario, SIP Server did not generate EventPartyDeleted
. (ER# 285845167)
SIP Server no longer drops a call if during a ringback tone treatment, it receives a 302 Moved Temporarily
message from a transferred destination. SIP Server now continues to process the call. (ER# 286319633)
In a scenario where a call comes through a Trunk
DN that has the value of the refer-enable
option set
to false
and the call is routed to a Trunk Group
DN, after the switchover the new primary SIP Server no longer sends a REFER request, while processing the REFER request that was issued by the Trunk Group
DN. (ER# 286392900)
If SIP Server does not receive an answer SDP for the offer SDP in a re-INVITE message, SIP Server now resets the SDP state to let other operations continue within the same SIP dialog. Previously, in this scenario, SIP Server incorrectly left the call in the invalid state, so no operation with the SIP dialog of the call was possible. (ER# 287014753)
In multi-site call-monitoring scenarios, SIP Server now correctly generates an EventPartyDeleted
message when a caller is disconnected after the conference completion with an agent at the remote site. Previously, in this scenario, SIP Server did not generate EventPartyDeleted
even though it generated EventPartyAdded
. (ER# 289293243)
SIP Server no longer prints a duplicate total_timeout timer set
message into the log. (ER# 290416312)
SIP Server now correctly applies the start_timeout
timer if the USER_ANN_ID
and USER_ID
parameters are included in a PlayAnnouncementAndDigits
treatment request. Previously, in this scenario, SIP Server did not apply the start_timeout
. (ER# 290509215) [Fixed Known Issue]
SIP Server now successfully applies the dial plan to a call during processing of a TRedirectCall
request. Previously, SIP Server did not apply the dial plan to a call and failed to redirect the call to an external destination. (ER# 291105539)
When the graceful shutdown command is issued, SIP Server now performs graceful shut down by transiting to the suspended state, and releasing the calls. Previously, instead of graceful shutdown, SIP Server restarted as it was defined (Auto-restart) on the Start Info tab in the SIP Server Application. (ER# 291663084)
SIP Server now correctly recovers recording and conference sessions if MCP goes down. Previously, SIP Server was not able to recover the service on a second Media Server. (ER# 292439093)
When the overload control feature is enabled while SIP Server is running, the dialog rate is now calculated from the time when the overload control feature is enabled. Previously, in this scenario, the dialog rate was calculated from the start of SIP Server, which led to calls being rejected because of the high dialog rate. (ER# 293054179)
When processing a TMakePredictiveCall
request, SIP Server now sends the Expires
header containing appropriate values in the INVITE message sent via an alternate trunk while the first selected trunk was unresponsive. Previously, SIP Server included incorrect values in the Expires
header. (ER# 294647484)
SIP Server now correctly sends an ACK
message in response to a 200 OK
message from Media Server when the inbound leg of the call is re-INVITEd and then terminated during the MSML-based treatment. Previously, in this scenario, SIP Server did not send the ACK
message and attempted to initiate a new treatment to the call. (ER# 295442946)
If a DN is disabled and then deleted from the configuration environment, and SIP Server receives a TRegisterAddress
request for this DN, it now generates an EventError
message. Previously, SIP Server incorrectly processed registration for the deleted DN. (ER# 295771299)
SIP Server now correctly handles a scenario where a transfer destination is released from a call when a 3pcc Complete Transfer operation is in a progress. SIP Server now keeps the main call active. The problem occurred when the transfer controller DN was configured with the dual-dialog-enabled
option set to false
. Previously, in this scenario, the main call became non-operable, and SIP Server responded to any T-Library request with EventError
. (ER# 296043909)
In a multi-site scenario, SIP Server now plays greetings after completing TCompleteTransfer/TCompleteConference
requests when the greetings-after-merge
parameter is enabled. Previously, in this scenario, SIP Server did not play greetings. (ER# 296342146)
In a call-monitoring scenario with MonitorScope
set to agent
, where a supervisor issues a TCancelMonitoring
request and an agent performs a single-step transfer, SIP Server now correctly removes the supervisor from the call. Previously, when the supervisor canceled the monitoring session, SIP Server did not remove the supervisor from the call. (ER# 296566076)
The SIP Server Application TServer_SIPPremise_810.xml
file no longer contains duplicate entries of wrap-up-time
and untimed-wrap-up-value
configuration options. Previously, duplicate entries caused conflict in Genesys Administrator. (ER# 296586439)
When playing a ringback tone, SIP Server now correctly selects a Voice over IP Service
DN with the geo-location of the agent who initiated the TMakeCall
request. Previously, when playing a ringback tone, SIP Server incorrectly selected the Voice over IP Service
DN with the geo-location of the destination DN. (ER# 297747182)
In a call-monitoring scenario, if greeting and monitoring are enabled for an agent but the greeting fails, SIP Server correctly starts monitoring. Previously, in this scenario, SIP Server did not start monitoring. (ER# 298322594)
After a session refresh, SIP Server now correctly processes consecutive 3pcc requests. Previously, after a session refresh, if SIP Server received a re-INVITE request from a caller, SIP Server did not process it and responded with a 491 Request Pending
message. (ER# 298519034)
SIP Server is now able to play customer or agent greetings that are requested in the TRouteCalll
request sent to the remote site. Previously, these greetings were sometimes not played if the access resource number provided by the remote site was resolved to one of the available trunks, and, at the same time, the access resource name matched exactly the name of the configured but disabled trunk on the SIP Server switch. (ER# 298860730)
When handling event distribution, SIP Server no longer becomes unstable when a device with an active call, but without registered T-Library clients, was deleted from the configuration environment. (ER# 298943328)
SIP Server no longer becomes unstable in a scenario where the time specified in the agent-no-answer-timeout
option expires and SIP Server redirects a call from the agent's DN (configured with reuse-sdp-on-reinvite=true)
to overflow destinations sending a REFER message only to the dialog which is fully completed. Previously, SIP Server sometimes sent a REFER message to the dialog that had an non-acknowledged re-INVITE transaction. (ER# 299261303)
If, when applying a treatment to a call, SIP Server gets a 488
error response from a gateway, SIP Server now parks the call once and does not attempt to re-apply the treatment. Previously, SIP Server attempted to park the call each time it received the 488
error message, which led to an infinite loop of the message exchange between SIP Server and the gateway. (ER# 299843510)
When SIP Server functions as an application server behind a softswitch and the softswitch responds with a 408 Request Timeout
message, SIP Server now correctly places a DN in out-of-service state. After the recovery-timeout
(configured on the Voice over IP Service
DN with service-type=softswitch
) setting expires, SIP Server places the DN back in service. Previously, in this scenario, SIP Server did not place the DN back in service. (ER# 300521299, 300606487)
In a multi-site scenario where a call is routed from Site A to Site B using the ISCC route
transaction type, and the greeting parameters are specified in the TRouteCall
request, the agent greeting is only played by SIP Server at Site B. Previously, in this scenario, SIP Server at Site A and SIP Server at Site B simultaneously engaged Stream Managers to play the greeting. (ER# 300633942, 303202526)
SIP Server now correctly prints the total Trunk capacity in the log. Previously, SIP Server printed the incorrect value of the total capacity of the Trunk, which had the capacity-group
option configured but not the capacity
option. (ER# 300638934)
SIP Server no longer treats inbound calls as business calls in multi-site scenarios when the option propagated-call-type
is set to true
and the option internal-bsns-calls
is set to false
. This means that the agent does not go into the AgentNotReady state after the call is released. The issue occurred when the Switch Partitioning functionality was enabled (the dn-scope
option was set to tenant
).
Limitations
PropagatedCallType
feature is only supported in Primary SIP Server. The backup SIP Server in an HA pair is not aware of the PropagatedCallType
, so after an HA switchover the PropagatedCallType
may be changed.
(ER# 301478945)
When processing a TRouteCall (RouteTypeDefault)
request that requires delivering a call to a default destination, SIP Server checks for the default-dn
configuration option that might be specified in the following locations, in order of priority:
default-dn
destination specified at the Application level. (ER# 301789073)
SIP Server now properly handles a User-Agent
Extension attribute in TRouteCall
requests. Previously, SIP Server incorrectly added duplicated User-Agent
headers when sending INVITE messages. (ER# 301864233)
SIP Server no longer cancels the dialog with MSML-based services to apply a treatment when it receives a PRACK message for an inbound leg of the call after the msml
dialog is initiated in response to a TApplyTreatment
request. (ER# 302221846)
If the value of the sip-ring-tone-mode
configuration option is set to 1
, SIP Server no longer simultaneously plays a ringtone and a treatment that was already applied to a call. (ER# 302446624)
For NETANN-based services with Genesys Media Server, SIP Server passes received digits in the INFO message without any conversion to the code (as it does with Stream Manager). Now SIP Server does not convert the non-digit characters *
and #
to DTMF(0). (ER# 303202549)
In a multi-site scenario, SIP Server now correctly sets the CallState
attribute to OK (0)
in EventQueued
messages if it receives a 302 Moved Temporarily
message from another SIP Server after a multi-site route. Previously, in this scenario, SIP Server incorrectly set the CallState
to Forwarded (23)
. (ER# 304149137)
In a multi-site call-monitoring scenario where:
MonitorMode
is set to mute
,TMonitorNextCall
request with MonitorMode
set to coach
for the same agent.TMonitorNextCall
request from the supervisor. Previously, in this scenario, SIP Server rejected the TMonitorNextCall
request. (ER# 304510594)
SIP Server no longer rejects 3pcc requests containing blank spaces in destination DN names.It executes the 3pcc requests by truncating the blank spaces. (ER# 307294336)
When, after a single-step conference of a call to a Routing Point while the treatment is in progress, a caller is disconnected from the call, SIP Server correctly processes an incoming TRouteCall
request with RejectRouteType
and disconnects parties from the call. Previously, SIP Server generated an EventError
message. (ER# 308070301)
If the value of the sip-enable-sdp-codec-filter
configuration option is set to true
and SIP Server processes multi-part content SDPs, SIP Server now correctly filters SDPs based on the specified codecs. Previously, in this scenario, SIP Server did not filter the codecs correctly, which resulted in dropped calls. (ER# 308232199)
SIP Server now correctly exchanges session re-INIVTE messages while processing a call with a treatment. Previously, because of network issues, SIP Server did not process the call correctly, which resulted in a stuck call. (ER# 308509325)
SIP Server no longer rejects a call containing the Content-Type=multipart/mixed
in an INVITE request. Previously, SIP Server generated an EventError
message. (ER# 309203586)
In a predictive-call scenario involving a treatment by NETANN-based services, SIP Server no longer rejects the second TApplyTreatment
request if the NETANN-based service replies to the first TApplyTreatment
request with the BYE message. (ER# 310815935)
A primary SIP Server now correctly synchronizes RTP monitoring information with the backup SIP Server, even if the backup SIP Server was not running when the primary SIP Server received a subscription (TPrivateService
request) for RTP monitoring information. As a result, after the backup SIP Server becomes the new primary, it is able to distribute RTP monitoring information in the EventUserEvent
message. Previously, SIP Server synchronized the RTP monitoring information only if the backup SIP Server was running. (ER# 311006592)
SIP Server no longer drops a consultation call to an external destination if the external destination responds with a 302 Moved Temporarily
message while playing a ringback tone to a transfer control party. (ER# 311629789)
SIP Server now correctly generates an EventError
message if a single-step conference is made to an unresponsive DN. Previously, SIP Server correctly terminated conference dialogs and reconnected the caller with an agent, but it did not generate EventError
. (ER# 311878826)
SIP Server no longer terminates a call in a scenario involving a ringback tone on an endpoint and early media on a destination. (ER# 312023908)
SIP Server now attaches the correct UserData if the Content-Type
is application/vnd.radisys.msml+xml
and the header in the INFO message contains the data in non-UTF-8 format. Previously, SIP Server converted the data to the UTF-8 format twice, and as a result, attached the incorrect UserData to the call. (ER# 313635841)
SIP Server now correctly forwards an INVITE message with the UserData to a Media Server when it completes a transfer while the treatment (NETTANN-based) is in progress by processing the INVITE message with the Replaces
header. Previously, in this scenario, SIP Server did not attach the UserData when sent the request to Media Server. (ER# 314294955)
In an Outbound VOIP environment, when processing TMergeCalls
with the Extension attribute method
set to bridging
, SIP Server now correctly bridges the call legs. Previously, SIP Server after the bridging immediately disconnected the call legs and sent the re-INVITEs to both call participants to restore the call, causing the delay in connection of the call participants. (ER# 314972576)
SIP Server now, when sending a 200 OK
response for the established dialog, includes all supported request methods (INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO, UPDATE, MESSAGE, NOTIFY, OPTIONS) in the Allow
header to a media gateway. Previously, SIP Server included only a limited set of request methods as in the previous 18x
provisional response. (ER# 315060681)
SIP Server now escapes the reserved character =
in the URI parameters of an INVITE message that is received in the form dn=msml=
for call recording. Previously, SIP Server did not escape the reserved character that caused incorrect call handling by GVP. (ER# 315202909)
SIP Server now terminates a call leg established with Media Server (for playing a ringback tone) after the call is established with the agent using a dummy SDP. Previously, in this scenario, SIP Server dropped the call. (ER# 315951629)
SIP Server now correctly processes a scenario in which the ringback tone (sip-ring-tone-mode=1
) to be played to a caller fails. SIP Server now keeps the call until the agent answers. Previously, when a Media Server responded with an error, SIP Server sent a BYE message to the caller and a CANCEL message to the agent leg. (ER# 283915697, 316899513)
When a two-step conference is successfully completed and SIP Server sends an INFO message to the MCP but receives an error response instead, SIP Server now correctly reports the DN of the agent who initiated the conference, as the value of the ThirdPartyDN
attribute in the EventPartyAdded
message. Previously, SIP Server incorrectly reported a caller DN as the ThirdPartyDN
attribute value. (ER# 317241516)
SIP Server now updates the contact of the SIP endpoint in the configuration environment only if it is different from the contact received in the previous REGISTER message. Previously, SIP Server updated the contact each time the endpoint sent the refresh REGISTER message, which led to minor usability issues. (ER# 317779268)
If a DN subscribes to a message-summary
event, but the DN has no mailbox configured (option gvm_mailbox
), SIP Server accepts the subscription and no longer sends a NOTIFY
message. (ER# 299401974)
When sending an outbound INVITE request, SIP Server will generate a P-Asserted-Identity header from the P-Asserted-Identity header that was included in the incoming INVITE if the enforce-privacy
option is not configured on the outbound Trunk DN. Previously, in this scenario, SIP Server sometimes did not generate a P-Asserted-Identity header. (SIP-14559)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.68. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer terminates unexpectedly after receiving a T-Library message that contains a corrupted/malformed TKVList pair in AttributeUserData, and now, in this case, will reply with EventError. (SIP-13670)
SIP Server no longer terminates unexpectedly after a switchover from backup to primary mode if a call is diverted from a Routing Point to another DN when the after-routing-timeout
expires. Previously, the call was not properly diverted from the Routing Point, causing SIP Server to terminate after the switchover. (SIP-6622)
SIP Server no longer terminates unexpectedly if the destination SIP Server in an ISCC transaction receives two or more events propagated by ISCC followed by ISCCEventCallDeleted while the transaction is still in progress. Previously, SIP Server sometimes entered into a loop, consuming 100% of the CPU and triggering an Out of Memory error. (SIP-6621)
If an agent with a nailed-up connection is involved in main and consultation calls that are placed on hold, and if during an attempt to retrieve a consultation call the second party releases the consultation call, SIP Server now correctly releases the consultation call and places the agent back on the main call, which remains on hold. Previously, in this scenario, the agent was not re-invited to the main call. (SIP-6611)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.61. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In early media scenarios, SIP Server no longer starts call monitoring until a call is established between a caller and an agent. Previously, SIP Server started call monitoring immediately after Hold and Retrieve operations and before the call was established. (ER# 320474711)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.61. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer grows in memory in the ISCC Call Overflow scenarios.
Previously, when a call was routed by ISCC using the route
transaction type and the Call Overflow feature was activated, SIP Server
grew in memory. (ER# 321694858)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.61. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
For Delta-Proxy IMS deployments only: SIP Server no longer sets the orig
parameter in the Route header of INVITE messages going to an agent in predictive-call scenarios. Previously, in these scenarios, the orig
parameter was set in the INVITE messages going to both a customer and an agent. (ER# 314313212)
In predictive-call scenarios, when Call Progress Analysis (CPA) is in progress and SIP Server receives a BYE message from the outbound Trunk, SIP Server now generates an EventReleased message with the CallState attribute set to 5
(remote release) and GSW_CONNECT_TIME in the AttributeExtension. Previously, SIP Server generated EventReleased with CallState set to 0
. (ER# 320063213)
Supported Operating Systems
New in This Release
Corrections and Modifications
The operating systems supported by this release are listed in the Contents, above.
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.61. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
If, while in a conference, an agent issues TReleaseCall and TDeleteFromConference requests simultaneously, SIP Server now releases the agent from the call. Previously, in this scenario, when running in the multi-threaded mode, SIP Server terminated unexpectedly. (ER# 320766840)
SIP Server now supports version 11.9 of the FLEXlm client library, which is fully compatible with the previous version 9.5 of the FLEXlm license manager. Previously, this incompatibility sometimes caused SIP Server to terminate unexpectedly. (ER# 315850077)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.57. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server in high-availability mode, when integrated with Cisco Unified Communications Manager 7.1, now correctly synchronizes the agent state after a switchover occurs. Previously, if an agent was set to the Logout or NotReady state, after a switchover SIP Server erroneously set the agent back to the Login/Ready state. (ER# 320478391)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.57. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
In addition to the current supported Advice of Charge (AoC) feature where SIP Server uses the INFO method to send the AoC information, SIP Server can now send the AoC information using 183 Session in Progress
reliable provisional responses for non-established dialogs of inbound calls.
To enable this feature, the sip-enable-100rel
option must be set to true
at the origination device (Trunk) or at the SIP Server Application level. Setting this option on the device (Trunk), if configured explicitly, takes precedence over the Application-level option. SIP Server receives the AoC information in a TPrivateService
message sent from a T-Library client for a non-established dialog. SIP Server maps the AoC information from the
TPrivateService
to an 183
message instead of the INFO message. If AoC is successfully delivered, then SIP Server responds with EventACK. Otherwise, it responds with EventError.
Note: Currently, SIP Server does not enable the feature if the DN-level sip-enable-100rel
option is set to true
but the Application-level sip-enable-100rel
option is set to false
.
(ER# 318959353)
SIP Server now supports the BroadWorks SIP Trunking solution by using the Network Asserted Identity mechanism for controlling the presentation of personal information (caller details) in SIP messages within a trusted network. The following SIP headers are included when SIP Server generates outgoing SIP INVITE messages:
P-Asserted-Identity
header must contain the Address of Record (AOR) of the trunk. The header must be present regardless of the privacy settings. The value must be explicitly configured on a Trunk level.From
header must contain a valid identity of a caller known to both the Carrier's SIP Network and SIP Server, even when CLIR (anonymous call) is requested. This identity could be a specific user, or a number/identity associated with an outbound campaign or a routing point.Privacy
header must be present containing the option-specified value (user,id
if the recommended configuration is used).To configure this feature, set the following configuration options.
SIP Server Application Object:enable-preserve-privacy=true
network-provided-privacy=user,id
enforce-p-asserted-identity
=<Address of Record (AOR) of the trunk>enforce-trusted=true
TServer
false
true, false
When set to true
, the privacy service requested by an initial call is preserved during routing, transfer, consultation, and conference operations. When set to false
, those operations reset the call privacy state, unless it is explicitly requested by the origination DN.
TServer
user,id
DN-level option for Trunk
DNs. If configured, all outgoing INVITE messages sent through the trunk are altered in the following way:
From
header of an incoming INVITE request contains the valid caller ID information, it will not be replaced with an anonymous tag, even if privacy is requested. From
header contains an anonymous content and the P-Asserted-Identity
header is present, SIP Server replaces the anonymous value of the From
header with the P-Asserted-Identity
header content.From
header contains an anonymous content and the P-Asserted-Identity
header is not present, SIP Server passes the From
header with the anonymous content. From
header contains the caller DN information, it is altered if the CPNDigits override is requested.Privacy
header, if requested, will be overridden with the option value.P-Asserted-Identity
header will contain the value of the enforce-p-asserted-identity
option, if it is configured on the Trunk DN.Feature Limitations:
P-Preferred-Identity
header of the incoming INVITE message will not be taken into account to resolve the caller ID information. Privacy
header of the incoming INVITE and corresponding configuration options privacy
and enforce-privacy
, which SIP Server accepts, is id
. Any other value will not be interpreted as a privacy request. override-domain-from
option has no effect on an outbound Trunk, if a call is initiated by an incoming anonymous INVITE with the P-Asserted-Identity
header containing a real caller ID. (ER# 321020987) privacy
option will not be taken into account for routed calls. (ER# 320382287)P-Asserted-Identity header
will always contain the value of the destination trunk option enforce-p-asserted-identity
and the origination DN option p-asserted-identity
has no effect on the header value.(ER# 318959358)
This release includes the following corrections and modifications:
The backup SIP Server no longer terminates during the creation of 16777215th call if the sip-link-type
option is set to 0
.
(ER# 319151591)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.57. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly sets the AttributeReferenceID to the EventRouteUsed in the following multi-site scenario (the route
ISCC transaction type):
Previously, in this scenario, the EventRouteUsed message was generated, but did not contain the AttributeReferenceID. (ER# 315293468)
SIP Server now correctly deletes old log files according to the value set in the expire
option setting. Previously, logs were not removed when the value of the expire
option was set to <number
> of days. (ER# 318288685)
SIP Server no longer terminates a TRouteCall operation when a DTMF event is sent during a multi-prompt, interruptible treatment. Previously, in this scenario, SIP Server could incorrectly start to execute a new prompt and terminate routing. (ER# 304489007)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly releases the nailed-up connection if the registration expires or if the device is unregistered, placing a corresponding DN in out-of-service
status. Previously, SIP Server did not release the nailed-up connection for an out-of-service DN. (ER# 316589773)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly terminates a recording session of main and consultation calls if the parties that are left on the call after the call transfer do not have recording enabled. Previously, in this scenario, SIP Server erroneously continued a recording session after the recording-enabled party disconnected from the transferred call. (ER# 316425852)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer incorrectly includes the supervisor in the call when:
monitor-consult-calls
is set to false
.
SIP Server now deletes the START_TIMEOUT when the TOTAL_TIMEOUT expires before the START_TIMEOUT during PlayAndCollectDigit treatments when sip-treatment-continuous
mode is enabled. (ER# 314146737)
SIP Server now correctly generates an EventTreatmentEnd message when digits are received for TreatmentPlayAnnouncementAndDigits, even if the TreatmentPlayAnnouncementAndDigits is restarted because an INVITE with replaces for the original call is received (in other words, a consultation call transforms to a main call, 1pcc two-step transfer). Previously, SIP Server did not generate the EventTreatmentEnd message in this scenario. (ER# 315938599)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
This version was first released as a Hot Fix on 01/11/13.
This release includes the following corrections and modifications:
In outbound-call scenarios where early media is established between a called party and an agent, and the agent places the call on hold and then retrieves it, SIP Server now correctly reconnects the called party with the agent if the early media is converted to full media at the same time the call is retrieved (200 OK
has been received on the initial INVITE from the called party). Previously, this scenario caused a race condition, which resulted in a dropped call. (ER# 314843156)
SIP Server now restores a subscription to Media Server when it is promoted to primary SIP Server in scenarios where multiple switchovers occur. Previously, in these scenarios, Media Server was not available because SIP Server did not send a SUBSCRIBE message to Resource Manager unless out-of-service detection was configured. (ER# 315825434)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Support for EventPrivateInfo when integrating with Microsoft Lync. SIP Server generates an EventPrivateInfo message with corresponding MessageID (4021) to subscribed T-Library clients when an agent answers an incoming call with a TAnswerCall request. A T-Library client subscribes to this functionality by sending to SIP Server a TPrivateService request that contains the AttributePrivateMsgID = 3021. (ER# 314088183)
Support of Answer-Mode
SIP header in ''Auto'' mode as described in RFC 5373; compatible with Avaya 96xx phones. A new sip-answer-mode
configuration option must be set to Auto
to enable this functionality. This option can be set at both DN and Application levels. Setting at a DN level takes precedence over setting at an Application level.
When the feature is enabled, SIP Server adds an Answer-Mode
header with a value of Auto
into INVITE messages.
sip-answer-mode
Section: TServer
Default Value: An empty string
Valid Values: Auto
Changes Take Effect: Immediately
Specifies the content to be added to the Answer-Mode
header that is used to trigger the auto-answer functionality in the destination endpoint. SIP Server sends this header regardless of whether an endpoint has advertised support for the ''answermode'' sip.extension in the contact of a REGISTER message.
If this option is configured, SIP Server includes the Answer-Mode
header with the value of this option whenever it sends an initial INVITE message.
Note: Avaya phones send INVITE messages without a Referred-By
header in response to REFER from SIP Server; therefore, the refer-enabled
configuration option must be set to false
. Also, for Avaya phones, the dual-dialog-enabled
configuration option must be set to true
and the sip-cti-control
configuration option should not be configured.
(ER# 314008111)
This release does not contain any corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contain no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now distributes the correct AttributeCallState in EventReleased when GVP makes a 1pcc transfer (REFER) to a Routing Point with the X-Genesys-CallInfo
header set to routed
. (ER# 313675500)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Support for REFER authentication. SIP Server now supports the authentication procedure for outgoing REFER requests in case of 401 Unauthorized
or 407 Proxy Authentication Required
responses that contain the Authenticate
response header. See known limitations: ER# 312490411 and 312768969.
To configure authentication credentials on a DN of type Trunk, use the username
and password
configuration options:
username
Section: TServer
Default Value: An empty string
Valid Values: Any string
Changes Take Effect: Immediately
This DN-level configuration option specifies the username to be used in generating a response to the Digest challenge on this trunk. This option must be configured in the AuthClient
section on the DN Annex tab.
password
Section: TServer
Default Value: An empty string
Valid Values: Any string
Changes Take Effect: Immediately
This DN-level configuration option specifies the password to be used in generating a response to the Digest challenge on this trunk. This option must be configured in the AuthClient
section on the DN Annex tab.
(ER# 313124583)
This release includes the following corrections and modifications:
When operating in Disaster Recovery mode, in a scenario where a SIP registration expires on an agent DN that is configured to operate in dr-forward=no-agent
mode and is currently on the call, SIP Server logs the agent out after the call is released, and now sends the expected EventDNOutOfService
message. Previously, SIP Server did not send EventDNOutOfService
. (ER# 279059731)
SIP Server no longer supports the rq-expire-tout
and rq-expire-tmout
configuration options. Previously, if rq-expire-tout
was set to a value other than 0
, SIP Server did not send the EventLinkConnected
message. (ER# 312192909)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Improved logging for multithreaded-related issues. (ER# 308988285)
Thread-Alive Monitoring. SIP Server no longer loses the connection between main and transport threads due to the reason "connection too slow". SIP Server is now able to monitor activity on the transport and call manager threads. If SIP Server does not receive a response during ping operations from one thread to its opposite, it prints a message to the log. In case of delays from the transport thread, SIP Server issues warnings with the message ID 52031
. For delays from transport thread clients (Call Manager threads), SIP Server uses the message ID 52032
. Use these message IDs to configure alarm conditions or reactions. (ER# 312594883)
This release includes the following corrections and modifications:
In multi-prompt treatment scenarios, SIP server no longer incorrectly starts the second treatment prompt when the first prompt is finished but while the routing call operation by REFER method is still in progress. Previously, SIP Server started the second prompt before the routing was finished, causing routing to fail. (ER# 311917136)
SIP Server no longer terminates while processing an invalid Instant Messaging scenario, which previously led to a loop of INVITE messages. SIP Server now correctly clears the erroneously created dialogs. (ER# 311691238)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When SIP Server receives an 18x provisional message without the SDP, in response to it SIP Server now sends the PRACK message without the SDP. Previously, SIP Server incorrectly sent the PRACK message with the SDP. (ER# 311772128)
In a silent monitoring conference, a party that is retrieved from hold no longer hears the music-on-hold. Previously, the party continued to hear the music-on-hold after it was retrieved from hold. (ER# 310806976)
SIP Server now correctly processes a 200 OK message that is received in response to a re-INVITE request with a slight delay. Previously, in this scenario, SIP Server ignored the 200 OK message which led to a dropped call. (ER# 310940122)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When processing a Network Attended Transfer scenario, SIP Server now disconnects the media server legs for the music-on-hold treatment. Previously, in this scenario, SIP Server did not release MCP resources. (ER# 310816162)
When an outgoing INVITE request containing Privacy
and P-asserted-identity
headers is challenged with a 401 Unauthorized
response, SIP Server now correctly includes these headers in the re-INVITE for the authentication procedure. Previously, SIP Server included the Authorization
header but did not include Privacy
and P-asserted-identity
headers. (ER# 310516075)
In a nailed-up connection scenario where the use-register-for-service-state
option is set to true
, when the registration expiration timer expires and there is an active call, SIP Server now generates a DNOutOfService event once the active call leg is released (the BYE message is sent to that DN). Previously, in this scenario, SIP did not generate DNOutOfService event. (ER# 309869154)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
If a hold operation is issued during an agent or customer greeting, SIP Server now stops the greeting stream and processes the hold operation. Previously, SIP Server disconnected the call if the hold operation was requested during greetings. (ER# 276010650)
SIP Server now correctly initiates the start of music specified in the parking-music
parameter to a supervisor in case of remote monitoring, or an agent with the nailed-up connection. Previously, in the MSML-based configuration, neither the supervisor nor agent heard the music specified in the parking-music
parameter.
Limitation: If SIP Server receives an error message in response to the INFO message to start the parking music, the music will not be played.
(ER# 310364393)
If a call is transferred from one ACD Queue to another using the stranded call routing procedure, the caller now correctly listens to the default music while waiting in a queue. Previously, in this scenario, in the MSML-based configuration, the caller did not hear any music. (ER# 310262211)
SIP Server no longer terminates unexpectedly in a scenario where for a DN with a consultation call with recording Alternate Call and Release Call requests have been issued simultaneously, and then a 1pcc Hold Call request is attempted from the DN. Previously, in this scenario, SIP Server entered an infinite loop and terminated. To prevent this situation, SIP Server drops the consultation call in this scenario. (ER# 310827669)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In a call-monitoring scenario where the monitoring scope is agent
and a monitored agent makes a consultation call, the caller now correctly hears the music-on-hold treatment. Previously, in this scenario, the caller did not hear the music-on-hold. (ER# 309272832)
SIP Server now accepts the value of the Expires
header that is sent by an external Registrar. Previously, SIP Server ignored that value and sent re-REGISTER after the time specified in the Expires
header of the REGISTER request. (ER# 309823144)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now accepts and responds to an INVITE request even if the SIP
keyword is present in lower case in the Via
header. Previously, SIP Server ignored such a request. (ER# 309754089)
In a call-monitoring scenario where the monitoring scope is agent
, when establishing a consultation call from an agent, SIP Server now correctly connects the monitoring supervisor to the consultation call. Previously, in this scenario, the supervisor did not hear the consultation conversation. (ER# 308495707)
SIP Server no longer initiates the recovery-timeout
timer for a DN with no SIP registration. Previously, SIP Server could incorrectly place an unregistered DN back in service after the recovery-timeout
timer expired, in cases where the option use-register-for-service-state
was set to true
. (ER# 307201137)
If a re-INVITE message is received from a caller when an agent is still in the call establishing state, SIP Server now processes the call and then sends a re-INVITE for SDP re-negotiation. Previously, the re-INVITE message was processed before the agent dialog was connected, and the re-INVITE transaction failed. (ER# 308393687)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly updates the Contact
header and sends the INVITE message to the updated contact, which is provided in the outgoing REGISTER request. Previously, SIP Server sent the INVITE message with the updated Contact
header but it did not update the contact address in memory. (ER# 309218703)
After receiving an authentication challenge for a REGISTER request, SIP Server now sends the username
value, which is configured in the AuthClient
section, in the Authorization
header of the outgoing REGISTER message. Previously, SIP Server incorrectly sent the first part of the SIP URI from the force-register
configuration option. (ER# 308848894)
In a predictive-call-to-a-Routing-Point scenario and when the msml-support
configuration option is set to true
, SIP Server now correctly applies the PlayAnnouncementAndDigits
treatment. Previously, in this scenario, SIP Server attempted to use the same connection that was established with the Media Server (CPD leg) instead of establishing a new connection to the Media Server for playing the treatment. (ER# 308797548)
In a call-monitoring scenario where the monitoring scope is agent
and a supervisor cancels monitoring, SIP Server now correctly releases the supervisor from the monitored call after the switchover. Previously, in this scenario, SIP Server did not release the supervisor from the monitored call after the switchover. (ER# 308339463)
SIP Server no longer stops unexpectedly while trying to generate a TEvent for a party which was accidentally released while being on hold during transfer initiation. (ER# 316084941)
SIP Server no longer stops unexpectedly when the conference is initiated and the DN from which the conference initiated sends a BYE
message before the conference is established. (ER# 315053209)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In blind-transfer scenarios where an early media dialog is established with a consultation target, SIP Server now correctly connects the consultation target with a caller. Previously, in these scenarios, SIP Server did not connect the caller with the consultation target after the transferring party left the call. (ER# 306923218)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer processes DN state change synchronization requests that it receives when operating in Primary mode. Previously, some switchover scenarios may have caused SIP Server to incorrectly place certain DNs out of service because of mistimed synchronization requests. (ER# 304489053)
SIP Server now updates the contact
parameter in the 200 OK response for the REGISTER message when a modified REGISTER request is received. Previously, SIP Server did not update the contact
parameter while forming the response for the REGISTER refresh. Instead, SIP Server updated the contact
parameter only after the REGISTER request expired. (ER# 306403682)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
beepdetect
parameter in the msml
content of INFO messages to Media Server when SIP Server receives the am-beep-detection=on
Extension key in a TApplyTreatment request on the Routing Point, to which an answered outbound call has been transferred from a Trunk Group DN. Media Server will then wait for a beep from an answering machine before playing the treatment. (ER# 306923288)
This release includes the following corrections and modifications:
SIP Server now supports the restricted sip-wait-ack-timeout
configuration option, as follows:
TServer
When SIP Server processes an incoming re-INVITE request, it starts a timer to wait for the ACK message to be received for this transaction. Once the ACK arrives, SIP Server sends the re-INVITE to perform the requested operation (greeting, treatment, and so on). If the option is set to 0
(zero), the current SIP Server behavior applies.
Warning! Use this option only when requested by Genesys Customer Care.
Previously (or the current SIP Server behavior), when SIP Server sent the re-INVITE to perform the requested operation, it did not wait for the ACK message, which caused the call to drop. (ER# 306764542)
SIP Server now supports the keep-mute-after-conference
configuration option to control mute/unmute for the party when a conference call is released.
keep-mute-after-conference
Section: TServer
Default Value: false
Valid Values: false, true
Changes Take Effect: Immediately
When this option is set to true
and a third member of the conference is released, the muted party of the call must use the TSetMuteOff request to restore the voice path with a caller. When this option is set to false
, the voice path with a caller is restored automatically when a third member of the conference is released.
Previously, in a three-party conference, when a third member of the conference disconnected from the call, the caller was not able to hear the agent while the agent could hear the caller. (ER# 279765616)
SIP Server no longer terminates when attempting to resolve a situation in which a greeting is completed before the INVITE transaction has been acknowledged for the call that has recording enabled. (ER# 314605514)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This version was first released as a Hot Fix on 08/24/12.
This release contains the following new features or functionality:
Support for Geo-location in Active Call Recording. In Active Call Recording scenarios, SIP Server is now able to select the Media Server and Recording Server based on its geographic proximity to either the caller or the agent. This minimizes WAN traffic and telecom costs. SIP Server does not select the service itself, it passes the geo-location information (in the X-Genesys-geo-location
header) in the initial INVITE messages to Resource Manager, which then uses that information to select the closest Media Server to the caller or agent.
You can configure geo-location in any of the following places:
SIP Server selects and passes the X-Genesys-geo-location
header using a different order of configuration precedence, depending on the call scenario.
Inbound Calls
For inbound calls, the order of precedence for the geo-location configuration is:
Outbound Solution
For outbound calls using TMakePredictiveCall (ASM, proactive, or transfer mode), the order of precedence for the geo-location configuration is:
Outgoing Calls
For outgoing calls using TMakeCall (Agent makes a 3pcc or 1pcc outbound call to a media gateway Trunk DN), the order of precedence for the geo-location configuration is:
Limitation: Geo-location for call recording may not work in cases where multiple MSML media services are required.
(ER# 298681041)
This release includes the following corrections and modifications:
SIP Server now correctly clears the consultation call in cases where the reconnect call is started before the initiate transfer operation is completed. SIP Server now correctly handles the subsequent call after the main call is finished. (ER# 304624597)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In a multi-site intrusion monitoring scenario, SIP Server now correctly stops a supervisor monitoring session of the call that is single-step transferred to another destination. (ER# 306294299)
SIP Server now correctly sets the CallState to 0 at the EventQueued for the External Route Point for a call returning from another SIP Server by a TRouteCall request. Previously, SIP Server mistakenly set the CallState to 1 when, prior to the TRouteCall request, a TSingleStepTransfer request was issued by an agent. (ER# 304488903)
For a call that is queued on an ACD Queue, SIP Server plays the music file that is specified in the default-music
configuration option at the ACD Queue DN level. Previously, SIP Server ignored the value of this option and played the music file that was configured at the Application-level default-music
and/or music-in-queue-file
configuration options. (ER# 304354340)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Support for a new Extension, MonitoredDN
. SIP Server now includes the Extension MonitoredDN
in addition to Extensions MonitorScope
and MonitorMode
in the following events generated for the supervisor DN: EventRinging, EventEstablished, and EventMonitoring. (ER# 303327201)
This release includes the following corrections and modifications:
SIP Server no longer stops unexpectedly if, as SIP Server tries to play the ringback tone to the caller during a single-step transfer, the caller then releases the caller. (ER# 305166861)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly processes the scenario where two single-step transfers are initiated in a sequence within the same transaction. Previously, the second call transfer would fail if the greeting was played and the greeting-delay-events
was set to true
. The issue applied only to internal calls. (ER# 304461711)
In a multi-site call monitoring scenario, SIP Server now correctly stops a supervisor monitoring session of the call that is single-step transferred to another destination. (ER# 305012461)
SIP Server no longer terminates if Media Server becomes unresponsive in the scenario where an agent begins call recording and then parks the call by using a THoldCall
request, then the agent retrieves the call and gets an EventError
. Previously, SIP Server could terminate unexpectedly in such scenarios when the record-consult-calls
was set to true
and AttributeExtension record
was set to source
. (ER# 305024006)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.47. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Call Divert Destination. SIP Server supports routing the caller to a specific destination when, after an initial leg of the call is completed, only the caller remains on the line. For example, this feature could be used to route the caller to a post-call survey after the agent leaves the call.
To support this feature, SIP Server introduces a new DN-level option, after-call-divert-destination
.
TServer
Specifies the destination DN where SIP Server will divert the call in cases where the caller remains on the line when all other parties have left the call. For example, use this feature to send callers to an after call survey.
To enable this feature, configure this option on the Routing Point DN. You can also enable this feature by passing the after-call-divert-destination
parameter in the Extensions Attribute of a RequestRouteCall. Parameters passed in the Extensions Attribute override the value of the configured option.
Note: This feature is not supported for calls initiated by TMakePredictiveCall requests.
(ER# 300704365)
Mapping parameters for outbound calls. When SIP Server receives a TMakePredictiveCall from Outbound Contact Server (OCS), SIP Server can now pass customers parameters in the Extensions Attribute to the initial INVITE request that it sends to the customer. If these parameters already exist in the URI, SIP Server replaces these values with the new values from OCS. (ER# 300783011)
For Network Attended Transfer (NAT), SIP Server now supports both implicit and explicit transfers (including premature disconnection and blind transfer) as well as reconnect operations. The limitation requiring you to set the option sip-server-inter-trunk
to true
no longer applies. These NAT transfers now work for both settings of the option. (ER# 299625814)
This release includes the following corrections and modifications:
For NETANN-based services with Genesys Media Server, SIP Server passes received digits in the INFO message without any conversion to the code (as it does with Stream Manager). Now SIP Server does not convert the non-digit characters *
and #
to DTMF(0). (ER# 303202549)
SIP Server no longer treats inbound calls as business calls in multi-site scenarios when the option propagated-call-type
is set to true
and the option internal-bsns-calls
is set to false
. This means that the agent does not go into the AgentNotReady state after the call is released. The issue occurred when the Switch Partitioning functionality was enabled (the dn-scope
option was set to tenant
).
Limitations
PropagatedCallType
feature is only supported in Primary SIP Server. The backup SIP Server in an HA pair is not aware of the PropagatedCallType
, so after an HA switchover the PropagatedCallType
may be changed.
(ER# 301478945)
SIP Server now correctly processes the tel
scheme of the incoming Request-URI
in IMS configurations. (ER# 304631782)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.43. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
set-notready-on-busy
configuration option has been modified. In addition to true
and false
, the range of valid values now includes numbers or ranges of numbers separated by commas that represent SIP response codes.For example:
Setting | Description |
---|---|
set-notready-on-busy=486,408 | Not Ready state will be set for 486 or 408 responses only. |
set-notready-on-busy=true | Not Ready state will be set for all 4xx, 5xx, and 6xx responses. |
set-notready-on-busy=400-499 | Not Ready state will be set for any response code from 400 to 499. |
set-notready-on-busy=400-410,486,600-610 | Not Ready state will be set for any response code from 400 to 410, or 486, or codes from 600 to 610. |
(ER# 300704367)
This release includes the following corrections and modifications:
SIP Server no longer terminates when multiple devices with the Active Out-of-Service Detection feature remain Out-of-Service for significant periods of time. (ER# 301340135)
SIP Server now correctly processes responses from devices provisioned with identical Active Out-of-Service Detection feature parameters. Previously, SIP Server printed the SIP Message above is ignored
message to its log, and could stop processing the responses in high volume and frequency response scenarios. (ER# 301342181)
SIP Server now correctly places an unresponsive parked DN to out of service for a period of time configured in the recovery-timeout
option. Previously, SIP Server did not apply the recovery-timeout
option setting correctly for a nailed-up DN. (ER# 290191257)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.43. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
divert-on-ringing
configuration option at the DN level, for a DN of type Routing Point.
Additionally, a divert-on-ringing
key can now be used in AttributeExtensions
in TRouteCall
messages. (ER# 300704369)This release includes the following corrections and modifications:
If, when applying a treatment to a call, SIP Server gets a 488
error response from a gateway, SIP Server now parks the call once and does not attempt to re-apply the treatment. Previously, SIP Server attempted to park the call each time it received the 488
error message, which led to an infinite loop of the message exchange between SIP Server and the gateway. (ER# 302911776)
SIP Server now properly rejects an INVITE
and issues a 603
SIP error code when encountering malformed multi-part content. Previously, in this scenario, SIP Server could become unstable. (ER# 302195910)
SIP Server now correctly processes the location
key in AttributeExtensions
of TMonitorNextCall
requests in Business Continuity mode. Previously, SIP Server incorrectly processed TMonitorNextCall
requests if the monitoring site and the remote site had DNs configured with the same name. (ER# 302605112)
In the scenario where an agent issues the TRedirectCall
request while a fast busy tone is applied to a call, SIP Server now rejects TRedirectCall
and proceeds with playing the fast busy tone. Previously, SIP Server did not handle TRedirectCall
requests properly. (ER# 300470890)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.43. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly sets the To
header when operating in IMS mode, in cases where a TMakeCall
triggers a dial plan that contains the setting clir=on
. (ER# 301971908)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.43. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Detecting the type of MSML service. SIP Server supports new parameters in the Request-URI of the initial INVITE
to help identify the kind of MSML service being asked for, and for which tenant. (ER# 298681050)
This release includes the following corrections and modifications:
SIP Server no longer becomes unstable while attempting to process multiple TAgentLogin
requests that resulted in EventError
. (ER# 296404122)
SIP Server adds only one AttributeThisDN
to EventNetworkCallStatus
messages. Previously, SIP Server incorrectly added more than one AttributeThisDN
in those event messages. (ER# 300444585)
SIP Server no longer becomes unstable during the recovery of the conference, if the greeting recording and the conference are involved in the call. (ER# 296047737)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.43. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server can now be configured to include additional parameters in the Request-URI, in cases where the deployment requires it. For example, it can add the user=phone
in the Request-URI of INVITE
requests to a particular DN. To support this functionality, a new DN-level option has been introduced.
sip-uri-params
Section: TServer
Default Value: No default value
Valid Values: A string that contains valid URI parameters
Changes Take Effect: At next Call
Specifies which URI parameters SIP Server will add to the initial INVITE
request to start a dialog with this DN. If configured, SIP Server sends the specified parameters in the initial INVITE
to this DN. To include multiple parameters, enter in a semi-colon separated list.
(ER# 298681048)
This release includes the following corrections and modifications:
SIP Server is now able to suppress a busy tone when routing to an
agent using the direct-uui
transaction type. To support
this functionality, a new Application-level option,
enable-busy-on-routed-calls
, has been introduced.
enable-busy-on-routed-calls
Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: At next call
Specifies whether SIP Server plays a busy tone when routing calls to an
agent using the direct-uui
transaction type. If this option
is set to false
, the destination SIP Server does not play a
busy tone for routed calls to a busy agent in cases where the
route-type
is direct-uui
. SIP Server sends a
negative response to the route origination site, and the call can then be
routed to an available agent.
Previously, the destination SIP Server incorrectly played a busy tone to the customer and ended the call. This functionality is supported for Early Media call flows. (ER# 299626463)
SIP Server can now include the contact
header in unreliable
180 SIP messages. To support this functionality, a new Application-level
option, sip-add-contact-early-dialog
, has been added.
sip-add-contact-early-dialog
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: At next call
Specifies whether SIP Server adds the Contact
header to
unreliable 180
SIP messages. If you set this option to
true
, SIP Server adds the Contact
header to
unreliable 180
SIP messages. If you set this option to
false
, SIP server does not add the Contact
header to any unreliable 1xx
SIP messages (to provide
backward compatibility).
(ER# 299994154)
The metadata XML file has been updated. With this new XML file, you can now successfully deploy the SIP Server Application using Genesys Administrator. (ER# 298485141, 300387335)
With TSCP version 8.1.001.43, SIP Server now supports a new
Application-level configuration option, req-distrib-event-support
in the TServer
section, to resolve backward compatibility
issues.
req-distrib-event-support
Section: TServer
Default Value: no-check
Valid Values: no-check, check-client
Changes Take Effect: After SIP Server restart
This option controls the content of the T-Events, propagation of which
is requested by RequestDistributeEvent
. The scope of the
distribution depends on the presence and the values of
AttributeClientID
and AttributeReferenceID
, as
follows:
If the option is set to check-client
and if
AttributeClientID
is not present in the
RequestDistributeEvent
message, SIP Server does not include
AttributeReferenceID
in the event distributed to clients.
If AttributeClientID
is present in the
RequestDistributeEvent
message and a client is connected
with that ClientID
, SIP Server sends the event containing
AttributeReferenceID
to this client and sends the event
without AttributeReferenceID
to other clients.
If AttributeClientID
has a value of -2
and AttributeReferenceID
has a value of -2
,
SIP Server sends the event with AttributeClientID
and
AttributeReferenceID
to all clients registered for
ThisDN
. If AttributeClientID
has a value
of -2
and AttributeReferenceID
has a value
other than -2
, SIP Server sends the event without
AttributeReferenceID
to all clients registered for
ThisDN
.
If AttributeClientID
has a value other than
-2
and no client is connected with that
ClientID
, SIP Server sends an EventError
response to the client that sent the
RequestDistributeEvent
message and the requested event
is not distributed to clients.
If the option is set to no-check
, SIP Server does not
check for clients that are subscribed for
RequestDistributeEvent
messages and reports
AttributeReferenceID
in event messages (backward compatible
mode).
(ER# 296178791)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly handles UDP SIP traffic after receiving ICMP TTL Expired packets. Previously, after receiving such packets, the SIP Server UDP listener could become disabled. (ER# 296583034, 297510589)
SIP Server can now be configured so that Early Media does not affect the conversation between call participants when a Supervisor is monitoring the session. Previously, when a Supervisor initiated whisper coaching on an agent, the agent could inadvertently hear the ringing when the Supervisor was invited to the call.
To support this functionality, the following new configuration option was introduced.
call-observer-with-hold
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
If this option is set to the value true
, SIP Server sends the initial INVITE
with hold SDP to a Supervisor in a monitoring scenario. After the Supervisor answers the call, SIP Server sends a re-INVITE
to add the Supervisor to the conference.
(ER# 295610266)
SIP Server now includes the Max-Fowards
header in the ACK
messages it sends in response to a 491 Request Pending
message. (ER# 298548951)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Support for Server
and User-Agent
headers. SIP Server now supports inserting the Server
header into all replies that it sends, and the User-Agent
header into all requests. For the Server
header, you can configure this functionality at the Application level only. For the User-Agent
header, you can configure on either the Application or the DN level. For non-INVITE
dialogs, only the Application-level setting applies. You can also specify a User-Agent
Extension
attribute using the following T-Library requests: TMakeCall
, TMakePredictiveCall
, TSingleStepTransfer
, TSingleStepConference
, TInitiateTransfer
, and TInitiateConference
. Setting the User-Agent
using the Extension
Attribute overrides any values set in the configuration options.
You can use the following new configuration options to control this functionality:
sip-server-info$VERSION$, $APP-NAME$
*
Specifies whether SIP Server includes the Server
header in all reply messages that it sends. The default value for this option includes placeholders, which you can modify with your own information, as follows:
$VERSION$
= the build version
$APP-NAME$
= the name of the application in the environment
You can also use the special value *
, which specifies the Genesys SIP Server as the $APP-NAME$
and the current SIP Server build as the $VERSION$
.
$VERSION$, $APP-NAME$
*
User-Agent
header in all request messages that it sends. The default value for this option includes placeholders, which you can modify with your own information, as follows:
$VERSION$
= the build version
$APP-NAME$
= the name of the application in the environment
You can also use the special value *
, which specifies the Genesys SIP Server as the $APP-NAME$
and the current SIP Server build as the $VERSION$
.
(ER# 289974931)
This release includes the following corrections and modifications:
SIP Server now correctly reports the call party information (LCTParty
and LCTPartiesLength
) to its clients in multi-site single-step transfer scenarios involving a nailed-up connection.
(ER# 295688294)
SIP Server no longer incorrectly changes the UserData
after the call is routed, in cases where that UserData
was previously updated via a REFER
request and subsequently updated before the call was routed. Previously, SIP Server sometimes reset the UserData
to the old values used in the REFER
update. (ER# 298136582)
SIP Server now correctly forwards notifications to the DN subscribed for MWI irrespective of the 3pcc agent login or dynamic CONFIG update, in cases where mwi-notify-unregistered-dn
is set to true
and mwi-implicit-notify
is set to false
. (ER# 298838741)
SIP Server no longer terminates unexpectedly in scenarios where:
Extension
DN is not released.
TCompleteTransfer
.
(ER# 298800465)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.001.40. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In multi-site scenarios involving the pullback ISCC transaction type, SIP Server now distributes EventCallDeleted
and EventAbandoned
in the correct order in scenarios where the transaction expires at the origination site due to the call having already ended at the destination location of the transaction. (ER# 297522220)
In scenarios involving the pullback ISCC transaction type, SIP Server no longer allows ISCC transactions to expire in cases where a call is not found. (ER# 297344981)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.38. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature.
Enhanced MWI support. Using the new Application-level option mwi-notify-unregistered-dn
, SIP Server can now send NOTIFY
requests to the endpoint subscriptions regardless of the SIP registration for the endpoint.
mwi-notify-unregistered-dn
Section: TServer
Default Value: false
Valid Values true, false
Changes Take Effect: Immediately
Specifies whether SIP NOTIFY
requests are sent to an endpoint regardless of SIP registration. Required for integration with Genesys SIP Voicemail. If set to true
, SIP Server sends the MWI NOTIFY
to the phone even if the phone has not registered. If set to false
, SIP Server sends the MWI NOTIFY
to the phone only if the phone has registered with SIP Server.
Note: The above functionality works with the option mwi-implicit-notify
. This must to be set false
to support the above functionality.
This release includes the following corrections and modifications:
In a multi-site environment, SIP Server no longer plays a busy tone to customers while the call is being routed using the direct-uui
transaction type. To enable this behavior, a new Application-level option has been introduced.
enable-busy-on-routed-calls
Section: TServer
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
If set to false
, SIP Server does not play a busy tone for calls routed to a busy agent if the transaction type is direct-uui
. Instead, SIP Server sends a negative response to the route origination site and the call can then be routed to an available agent. If set to true
, SIP Server plays a busy tone to the customer and ends the call.
(ER# 296928921)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.38. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature.
Call Completion Features
SIP Server supports Call Completion on Busy Subscriber (CCBS) and Call Completion on No Reply (CCNR) when offered by a Siemens OpenScape Voice switch. This feature provides a callback mechanism, where a caller is able to request a call back from the switch when a line they have tried to reach (but is busy or does not answer) later becomes available.
This feature was successfully tested with Siemens OpenScape Voice version 6.0, patch number PS 20, E07.
Feature Configuration
SIP Server does not provide this functionality itself, but supports this functionality when offered by Siemens OpenScape Voice . To enable the feature, configure the following:
internal-registrar-enable
to false
.
external-registrar
to the same value as the contact
option configured on the softswitch DN (Voice over IP Service
DN with service-type
set to softswitch
).
With this configuration, SIP Server processes the Allow-Events: CCBS
and Allow-Events: CCNR
headers if included in the INVITE
request.
This functionality is available for 1 pcc calls only (not applicable for 3 pcc calls).
(ER# 288115313)
This release includes the following corrections and modifications:
SIP Server now correctly rejects calls in cases where a TRouteCall
request with a reject response after the previous TRouteCall
request results in an unanswered call. (ER# 295719848)
SIP Server now correctly removes the called party from the call when the after-routing-timeout
expires. This ensures that in conference scenarios, when after-routing-timeout
is set lower than the sip-invite-timeout
, SIP Server is able to successfully generate the EventPartyAdded
and complete the conference. (ER# 288242221)
SIP Server now correctly distributes updated user data in the EventRouteRequest
and EventQueued
messages in cases where the user data is updated before the TSingleStepTransfer
request to a Routing Point is issued. (ER# 296404170)
SIP Server now waits for the INFO
with an msml.dialog.exit
event in the Voice XML application dialog before generating an EventTreatmentEnd
message. Previously, SIP Server sometimes incorrectly sent the EventTreatmentEnd
on receiving the INFO
with the msml.dialog.exit
event in the Beep-detection dialog. (ER# 296645931)
SIP Server now correctly distributes the events in cases where a multi-site conference breaks down when the conference initiator leaves the call and the option epp-tout
is set to a value greater than zero. Previously, in this scenario, SIP Server sometimes did not distribute EventPartyChanged
. (ER# 295643896)
SIP Server now correctly applies call monitoring in multi-site scenarios. Previously, SIP Server sometimes applied call monitoring on a multi-site consultation call, even when call monitoring for consultation calls was disabled. The problem occurred when the Switch Partitioning functionality was enabled (the dn-scope
option was set to tenant
).
A new Application-level option, use-propagated-call-type
, has been introduced.
TServer
never
never, monitor
monitor
, SIP Server uses the call type defined at the origination site to identify whether to start monitoring. Genesys recommends using this option in environments where Switch Partitioning functionality is enabled. (ER# 296120909)
For supervisors in a nailed-up connection, SIP Sever no longer incorrectly sets up a mute mode for a new supervisor-monitoring session, in cases where the previous session was in the mute monitoring mode. (ER# 296799844)
In multi-site scenarios involving the pullback ISCC transaction type, SIP Server now correctly distributes the EventDiverted
, EventReleased
(on the abstract DN), and EventCallDeleted
messages in the proper order. Previously, SIP Server distributed EventReleased
on the abstract DN and EventCallDeleted
to clients earlier than EventDiverted
. This resulted in inaccurate historical call reporting. (ER# 296811773)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
There are no restrictions for this release. This release contains no new features or functionality.
This release includes the following corrections and modifications:
In cases where a second treatment is applied to a call, and this treatment includes a new geo-location tag, SIP Server now establishes a new dialog with the Media Sever DN using the new geo-location, and terminates the previous Media Server dialog. (ER# 294848371)
SIP Server no longer automatically cancels monitoring in cases where a supervisor drops the nailed-up connection. This behavior can now be controlled by using the new Application-level option, cancel-monitor-on-unpark
.
TServer
true
true, false
true
, SIP Server cancels the supervisor subscription when the supervisor is unparked. If set to false
, SIP Sever does not cancel the supervisor subscription in case of unpark. (ER# 292560970)
For supervisors in a nailed-up connection, SIP Sever no longer incorrectly sets up a mute mode for a new supervisor monitoring session, in cases where the previous session was of the mute monitoring mode. (ER# 294919048)
SIP Server now correctly sends re-INVITE
requests to the call parties in cases where the after-routing-timeout
expires but SIP Server has received a preview SDP. (ER# 287910463)
SIP Server now processes TRouteCall
requests in cases where a second MSML dialog for a TApplyTreamtment
request results in a BYE
message before the INFO
message can be processed. SIP Server also processes TRouteCall
requests in cases where the INFO
message in this scenario times out. (ER# 293394024)
SIP Server now correctly synchronizes the Do-Not-Disturb (DND) state of a DN to the backup SIP Server instance. (ER# 294848353)
When handling REGISTER
requests, SIP Server no longer adds extra characters to the IP address that represents the host machine that sent the request. Previously, SIP Server sometimes added extra characters to the via
header. (ER# 292514513)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
TNetworkConsult, TNetworkAlternate, TNetworkTransfer,
and TNetworkReconnect.
See the Framework 8.1 SIP Server Deployment Guide for details. (ER# 288115315)
This release does not include any corrections or modifications.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information about changes to the Common Part that might affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
When reject-call-incall
is set to true
on a Voiceover IP Service
DN (service-type=softswitch
),
if a second call is made to an Extension
DN behind this softswitch, SIP Server rejects the call with a 603 Decline
error response. Previously, SIP Server ignored this option when configured on a softswitch DN.(ER# 292536900)
SIP Server now handles gracefully 481 Call/Transaction Does Not Exist
error response sent by Media Server in response to a refresh re-INVITE
request that SIP Server sends after a switchover. Previously, SIP Server could terminate unexpectedly in this scenario.
(ER# 294271111)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly applies the value hide
to a key of the given KVList pair that is specified in the log-filter-data
section to filter data in the log. (ER# 293175065)
SIP Server now correctly closes file descriptors if the sip-address
configuration option specifies an address that is not present on the SIP Server host. Previously, SIP Server attempted unsuccessfully to open this non-existent port and sometimes left an unclosed file descriptor as a result. On the Linux server, this led to SIP Server sometimes consuming 100% CPU when the file descriptor limit was reached. (ER# 292802634)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
This release includes the following corrections and modifications:
When SIP Traffic Monitoring is configured, SIP Server initiates a switchover if no SIP messages are received during a period of time that SIP Server calculates as follows: SIP Server now adds 32
seconds to the sum of the two options oos-check
and oos-force
. The maximum value out of all configured DNs is used as the timeout for SIP Traffic Monitoring. (ER# 293008321)
SIP Server now successfully establishes multi-site conferences using the direct-uui
transaction type in cases where a Monitored agent accepts the conference through a PreviewInteraction
request. The Supervisor and Monitored agent are configured as nailed-up and parked. Previously, in this scenario, SIP Server incorrectly dropped the call when the agent accepted the request. (ER# 291911955)
SIP Server now properly releases the call in cases where the caller DN issues a CANCEL
before the agent is able to answer using the 3pcc TAnswerCall
request. Previously, in this scenario, race conditions sometimes resulted in a stuck call. (ER# 292796071)
When SIP Server receives an UPDATE
message, it now responds with a 200 OK
message. Previously, SIP Server could stop processing transactions in progress after receiving the UPDATE
message. (ER# 283638897)
SIP Server now correctly binds listeners to their respective IP address, as defined in the sip-interface
option. (See the option description in the Framework 8.1 SIP Server Integration Reference Manual.) In cases where a name is specified as the value for this option, SIP Server performs name resolution at application startup; the respective IP address is used whenever SIP Server binds a listener. If the IP address does not match any address configured on the host, or if the name resolution fails, SIP Server will not open the listener.
Note: Genesys recommends that you configure the sip-interface
option only in deployments where SIP Server high-availability is provided using the F5 BigIP Local Traffic Manager load balancer (in this case the values of the sip-interface
and sip-address
options will not match). (ER# 289870132)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.34. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
This release of SIP Server corrects the following TSCP issue:
SIP Server no longer becomes unresponsive while trying to reconnect to Configuration Server after connection to it was lost. (ER# 293261601, 292802634)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.31. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
SIP Server can now add the P-Early-Media
header in SIP messages that it sends to Genesys Media Server when applying a treatment on a Routing Point. Previously, a limitation of the option force-p-early-media
prevented SIP Server from adding P-Early-Media
headers in treatment scenarios involving Routing Points (see ER# 275228738 from Release 8.1.000.40). (ER# 290174029)
SIP Server can now suppress the To-Tag
parameter from appearing in REGISTER
requests. To support this functionality, a new DN-level option, force-register-disable-totag
, has been introduced.
force-register-disable-totag
Section: TServer
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Enables suppression of the To-Tag
parameter in REGISTER
requests. When set to true
, SIP Server does not add the To-Tag
parameter to any REGISTER
message that it sends.
(ER# 287389636)
This release includes the following corrections and modifications:
A new DN-level configuration option, sip-disable-greeting
, has been added to control which of the SIP Servers (in a multi-site environment) will play greetings.
sip-disable-greeting
Section: TServer
Default Value: false
Valid Values: false, true
Changes Take Effect: At the next established call
When this option is set to true on a Trunk
DN and SIP Server sends an outgoing INVITE
message to this trunk, the greeting is not started and the extension's greeting parameters are added to the outgoing INVITE
in a specific header. When this option is set to false, SIP Server behavior is not changed.
Previously, if one SIP Server played greetings and another SIP Server performed monitoring, it could result in race conditions.(ER# 281338351)
SIP Server no longer inadvertently adds the P-Early-Media
header to the 200 OK
message that it sends in response to an INVITE
. Only 200 OK
messages sent in response to a PRACK
or an UPDATE
message can contain the P-Early-Media
header. (ER# 290173981)
SIP Server now successfully processes the following scenario:
(ER# 288771197)
The override-to-on-divert
parameter now applies not only to Routing Point DNs and ACD Queue DNs, but also in the case of 1pcc single-step transfer using the REFER
method.
When override-to-on-divert
is set to true
, SIP Server overrides the username part of the To
header in the URI of an outgoing INVITE
message, replacing it with the username part of the Refer-to
header from the REFER
message.
When override-to-on-divert
is set to false
, SIP Server does not override the username part of the To
header in the URI of the outgoing INVITE
message; instead the username will remain the same as that of the party that initiated the single-step transfer (in other words, the originator of the REFER
request). (ER# 279203659)
This release of SIP Server corrects the following TSCP issue:
The minimum value of the periodic-check-tout
configuration option is now 10 seconds. Previously, there was no minimum value defined for this option. Values of 10 seconds or slightly greater can still cause a moderate deterioration of SIP Server performance. (ER# 291263131, 289225813, 289161653)
This release of SIP Server corrects the following TSCP issue:
In multi-site conference scenarios involving call path (trunk) optimization at one of the call locations, SIP Servers (or T-Servers) at other call locations now correctly generate EventPartyDeleted
messages when an external party leaves the conference. (ER# 291027891, 286608305, 286608345)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer incorrectly sends a 503 Service Unavailable
message in cases where the Session Description Protocol (SDP) in a re-INVITE
request, received during an uncompleted SIP transaction, has a static payload with undefined (or incorrectly defined) attribute values. (ER# 289980301)
SIP Server now correctly starts the start_timeout
timer for PlayAnnouncementAndDigits
treatments that are started in treatment-interruptible mode (option sip-treatment-dtmf-interuptable
set to true
). In this case, at the end of the final prompt in the treatment, if no input is received from a user, SIP Server will begin the start_timeout
timer. (ER# 289892051)
When installed on 32-bit RedHat operating systems, SIP Server now correctly sends updated Session Description Protocol (SDP) to the destination party in early media dialogs when the values of the session-id
and session-version
of the o=
parameter in the SDP are both 20-digit numeric strings. (ER# 286013645)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new feature and functionality:
SIP Server is now able to respond with a 503 Service Unavailable
error message in cases where Resource Manager is down and does not respond to the INVITE
request or SIP Server, by using the Active OOS Detection mechanism, places Resource Manager out of service. (Note that Active OOS Detection is not applicable to Trunk Group
DNs.)
The sip-error-conversion
option enables this functionality. For example, if this option is set to 0=503
and a Trunk or Trunk Group does not respond to the INVITE
request, SIP Server will respond with the 503
error message. If this option is set to 404=503
and a Trunk is placed out of service by the Active OOS Detection mechanism, SIP Server will respond with the 503
error message.
(ER# 288687091)
This release includes the following corrections and modifications:
SIP Server now supports the Preview Interactions feature for multi-site calls (ISCC routing) using the direct-uui
transaction type. Previously, the Preview Interactions feature was supported only for multi-site calls where the route
transaction type was used. (ER# 289447611)
SIP Server no longer generates the same CallUUID
attribute for two different calls in the following scenario:
dual-dialog-enabled
option set to false.
EventRinging
message for a new call, it re-used the CallUUID
attribute of the consultation call. (ER# 288777361)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features and functionality:
Algorithm of handling concurrent SIP requests between two communicating endpoints is improved. Conflict occurs if SIP Server receives an INVITE
message with the SDP, which cannot be delivered to the destination, due to the concurrent request. SIP Server now responds to the received INVITE
with the last known SDP from the peer, only if this SDP can be used as an answer for an SDP offer received in the incoming INVITE
. Otherwise, SIP Server responds to the INVITE
with the 503 Service Unavailable
message containing a Retry-After
header. SIP Server can change a direction attribute of the last known SDP, which it sends as an answer, to match it with the SDP offer received in the incoming INVITE
.
To further improve the conflict resolution algorithm, a new Application-level option, sip-491-passthrough
, is introduced.
sip-491-passthrough
Section: TServer
Default Value: false
Valid Values: true, false
Changed Take Effect: Immediately
Specifies whether SIP Server will forward 491 Request Pending
messages, sent in response to a re-INVITE
request, to the remote endpoint. This operation mode should be used in the environments where SIP message conflict resolution is preferred to be carried out by the endpoints and not by SIP Server.
(ER# 269119815, 269119976)
This release includes the following corrections and modifications:
SIP Server now sends an EventRouteUsed
message and drops the call if router-timeout expires and the default routing destination specified in the default-dn
option is out of service. Previously, SIP Server did not process such a scenario correctly, which resulted in a stuck call. (ER# 268617103)
In HA configurations, Emulated After Call Work (ACW) now works correctly in cases where a switchover occurs after the call was established. Previously, the backup mode SIP Server did not generate an EventEstablished
message, and so, after the switchover, the agent remained in the Ready
state when the call was released. (ER# 281427141)
In 1pcc hold transactions, SIP Server now correctly passes custom headers in 200 OK
to a hold controller, in cases where the Application-level option sip-enable-moh
is set to false
and no media server is involved. (ER# 278355561)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Call release tracking. SIP Server supports reporting the identity of which party (agent or customer) is responsible for releasing a particular call. (ER# 284079505)
This release includes the following corrections and modifications:
SIP Server now correctly handles Nailed-Up connections for Supervisors. If Supervisor does not answer a monitored call within the No-Answer-Timeout period, SIP Server does not establish the Nailed-Up connection (it drops the Supervisor from the call). Previously, SIP Server tried to park the Supervisor, causing the Nailed-Up connection to work incorrectly. (ER# 285334961, 285335024, 285335080)
SIP Sever now correctly applies the reject-call-in-call
option for parked agents in a Nailed-Up connection, in both single and multi-site deployments. (ER# 287389723)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
SIP Server now applies the START_TIMEOUT
parameter in PlayAnnouncementAndDigits
treatments after the prompt
is finished, in cases where the MSML protocol is used, and there is no input from caller to interrupt the treatment. If the caller does not enter any digits before the START_TIMEOUT
finishes, SIP Server sends an INFO
message to stop digit collection. Previously, in this scenario, SIP Server applied the START_TIMEOUT
parameter before the prompt started. (ER# 283976115)
SIP Server now correctly modifies the userpart
in the Refer-To
header irrespective of the value of the options override-domain-oosp
or override-domain
. Previously, SIP Server incorrectly modified the userpart
of the Refer-To
header when override-domain
was configured on the Trunk. Now, these override parameter settings modify only the host part of the Refer-To
header, while the userpart
is modified based on the replace-prefix
option.
(ER# 284411995)
SIP Server no longer sends two subsequent requests to Configuration Server when an Extension
DN updates its contact information using a REGISTER
message. Previously, when the option sip-preserve-contact
was set to true
, SIP Server erroneously sent two subsequent requests to Configuration Server to update the contact. (ER# 284811776)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features and functionality:
Disconnect on remote agent logout. SIP Server now releases a nailed up connection when the remote agent logs out. (ER# 284079503)
This release includes the following corrections and modifications:
SIP Server no longer becomes unstable if it receives an INVITE
with an incorrect value in the Contact-Length
header. Previously, an incorrect value for the Contact-Length
, which did not match the actual SDP length, resulted in exceptions during message parsing. (ER# 278772421)
SIP Server no longer plays a ringback tone when processing a single-step conference call. Previously, when the sip-ring-tone-mode
option was set to 1
, SIP Server sometimes incorrectly started to play the ringback tone in this scenario, and as a result, the third conference participant could not hear the other participants. (ER# 286013824)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features and functionality:
Enhanced SIP error code mapping. SIP Server supports mapping standard SIP and MSML errors sent by GVP to resulting T-Library messages. (ER# 284079501)
This release includes the following corrections and modifications:
SIP Server now correctly recognizes the end of a GVP treatment in the following scenario:
INFO
message requesting the end of the treatment, then sends the EventTreatmentEnd
.
INFO
message and did not end the treatment.
(ER# 282061473)
SIP Server now correctly parses the parameters in the Refer-To
header of incoming REFER
requests. Based on these parameters, SIP Server creates the correct Replaces
header in the new INVITE
to the transfer destination. Previously, parsing errors could cause SIP Server to send the wrong Replaces
header in these INVITE
requests, and as a result the transfer requested on the remote site failed.
This issue occurred in multi-site environments where the Trunk pointing to a remote SIP Server instance was configured with sip-server-inter-trunk
set to true
and sip-replaces-mode
set to 1
. (ER# 285802548)
SIP Server no longer incorrectly ends the call on receiving an UPDATE
message after an INVITE
request for a TMakePredictiveCall
request. Previously, in this scenario, upon receiving the UPDATE
message for a non-established dialog, SIP Server dropped the predictive call. (ER# 282263445)
In scenarios where a call is put on hold by the calling party while parked on a media server (Voice over IP Service
DN with service-type
set to Application
) for a treatment, SIP Server now successfully uses the re-INVITE
method to put the media stream on hold. Previously, this scenario could sometimes result in no media stream flowing after the call was retrieved from hold. (ER# 281366782)
SIP Server now correctly ends a monitored call attempt when the invited supervisor does not answer. Previously, SIP Server incorrectly routed the call. (ER# 268402681)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server HA pair now correctly synchronizes CallID
s if there are network delays during the primary SIP Server restart. Previously, network delays could cause race conditions, where both instances in the HA pair (primary and backup) were running in backup mode, resulting in all calls being released. (ER# 283266731)
If an incoming INVITE
for an inbound call contains the SIP Alert-Info
header, and the Alert-Info
header is also specified in either the DN or Application-level options, SIP Server now correctly adds a single Alert-Info
header in the INVITE
request that is sent to the endpoint. Previously, SIP Server added two Alert-Info
headers. (ER# 280130309)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
There are no restrictions for this release. This release contains the following new features or functionality:
Privacy
and P-Asserted-Identity
header values that are set on outbound Trunk
DNs in SIP messages, instead of the values set on inbound Trunk
DNs.TServer
id
Trunk
DN (instead of the privacy
option, which provides the privacy value set on an inbound Trunk
DN). This option is also applicable to DNs of type Extension
and Voice over IP Service
. This option is not used in COLR (Connected Line Identity Restriction) scenarios.TServer
P-Asserted-Identity
header. For Extension
DNs, it provides the P-Asserted-Identity
value equal to the DN. For CLIP/CLIR (Calling Line Identification Presentation/Calling Line Identification Restriction) scenarios, this option provides an identity value that should be set on an outbound Trunk
DN (instead of the p-asserted-identity
option, which provides the identity value set on an inbound Trunk
DN). This option is not used in COLP/COLR (Connected Line Identification Presentation/Connected Line Identity Restriction) scenarios.
enforce-p-asserted-identity
options are applied.enforce-privacy
and enforce-p-asserted-identity
options are applied.enforce-privacy
and enforce-p-asserted-identity
options are applied.privacy
and p-asserted-identity
options are applied.
GSIP_RECORD
key of UserData
in EventAttachedData
messages. The valid values of the GSIP_RECORD
key are ON, OFF,
and PAUSED
. When recording is established, the value of GSIP_RECORD
should be set to ON
. (ER# 278510359)
This release includes the following corrections and modifications:
SIP Server now correctly releases the transfer destination party in cases where a caller disconnects the call while a 3pcc CompleteTransfer operation is in progress and consultation call recording is enabled. (ER# 283085735)
SIP Server now properly generates voice monitoring events (EventUserEvent
) that indicate both the start and end of the conversation that takes place between two parties after a call transfer is completed using a TCompleteTransfer
request. Previously, in this scenario, SIP Server did not generate voice monitoring events correctly. Note that the start EventUserEvent
message is generated with the ConnID
of the consultation call, and the stop EventUserEvent
message is generated with the ConnID
of the main call. (ER# 283085668)
SIP Server Application Template files (TServer_SIPPremise.tpl
and TServer_SIPPremiser.apd
) now contain valid configuration options. (ER# 283018475)
SIP Server now correctly sets the GSIP_REC_FN
key in EventAttachedDataChanged
to the value of the record
parameter contained in the To
header of the INVITE
message. (ER# 275881162)
SIP Server no longer becomes unstable if the MSML treatment timeout occurs after the TRouteCall
request is issued. (ER# 282866004)
In a scenario where a call is made from DN1 to DN2 (DN1 and DN2 have the record
option set to true
) and then the call is transferred from DN1 to DN3 (DN3 has the record
option set to false
), SIP Server now correctly processes the call transfer using the recorder associated with DN3. Previously, in this scenario, SIP Server selected the recorder associated with DN1, which led to incorrect call parties reporting. ER# 265962622)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.25. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality:
Transfer-Type
key in AttributeExtensions
for TRouteCall
requests. The routing strategy can use this key to select the SIP call flow that will be used to deliver the call to a routing destination. The valid values of the Transfer-Type
key are as follows:
invite
—INVITE
transaction is used to connect an origination party with a routing destination.
refer
—REFER
method is used to complete the call routing. The Host
name in the Refer-To
header points to SIP Server. So, SIP Server remains in the signaling path when two parties are connected after the routing.
NOTE: The REFER
method cannot be used if the call is not answered by the remote party (for example, when an inbound call is under a treatment on a Routing Point).
oosp
(Out Of Signaling Path)—SIP Server is taking itself out of the signaling path by sending the REFER
request or 302
response, which points to the routing destination, to the origination party. Two parties are connected directly when routing is complete. SIP Server no longer controls the call.Transfer-Type
key with values refer
and invite
has a higher priority than the refer-enabled
option configured on a DN of type Trunk
or Extension
. And, the Transfer-Type
key with the value of oosp
has a higher priority than the oosp-transfer-enabled
option configured on a DN of type Trunk
. SIP Server always checks the endpoint capability to support REFER
. If an endpoint does not announce REFER
support through SIP messages in the Allow
header, the REFER
message will not be sent to this endpoint, even if it is requested in the Transfer-Type
key.
(ER# 276256314)
force-p-early-media
, has been added in this release, in the TServer
section:TServer
false
true, false
P-Early-Media
header into requests and responses, even if the P-Early-Media
header with the supported value is not present in the initial INVITE
message.
P-Early-Media
header in the following messages: 18x, 200, INVITE, PRACK,
and UPDATE
.
force-p-early-media
is set to true
), this feature does not apply to scenarios involving treatments on Routing Points.This release includes the following corrections and modifications:
This version of SIP Server is built with TSCP 8.1.000.25, which corrects the following issue:
HA T-Server or SIP Server no longer becomes unstable in an environment where a new application object is created with a connection to the running HA T-Server or SIP Server. This issue was applicable only to HA deployments, and could occur in any of the following scenarios:
After a switchover, when the new primary SIP Server gets a NOTIFY
message from GVP for a dialog created by the backup SIP Server (former primary), it will no longer match any dialogs created by the backup SIP Server, and it will send a 481 (Call/Transaction Does Not Exist)
error message, which will terminate the subscription at the GVP side. Previously, after a switchover, the subscription created by the primary SIP Server was not terminated and it remained for 10 minutes, unless GVP terminated the subscription after the subscription timer expired. (ER# 272893300)
SIP Server now supports a new Application-level option, bsns-call-dev-types
, which allows inclusion of External Routing Points to the list of devices to which business call handling is applied to. SIP Server now accepts the BusinessCallType
key of AttributeExtensions
in TRouteCall
requests. Previously, SIP Server excluded External Routing Points from the list of devices to which business call handling was applied, and as such did not accept the BusinessCallType
key.
bsns-call-dev-types
Section: TServer
Default Value: +acdq +rp +rpq +xrp
Valid Values: A set of space-separated flags:
+/-acdq |
Turns on/off the classification of the call type as business on an ACD Queue |
+/-rp |
Turns on/off the classification of the call type as business on a Routing Point |
+/-rpq |
Turns on/off the classification of the call type as business on a Routing Queue |
+/-xrp |
Turns on/off the classification of the call type as business on an External Routing Point |
-rp
, calls to Routing Point DNs will not be automatically classified as business, allowing the routing strategy to use the BusinessCallType
Extension.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.1.000.23. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
In a multi-site deployment, SIP Server now correctly routes a call to a nailed-up agent using the direct-uui
transaction type, even if the agent's device is configured with the reject-call-incall
configuration option set to true
. (ER# 280933991)
SIP Server now correctly encodes and decodes UserData special XML characters (such as & (ampersand), < > (angle brackets), " (double quotation mark), and ' (single quotation mark)) in MSML SIP messages. (ER# 281086259)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with the T-Server Common Part (TSCP) release number 8.1.000.23. TSCP is the shared software that all T-Servers and SIP Servers use. Consult the TSCP release note for information on changes to the Common Part that may affect the functionality of your particular type of T-Server.
There are no restrictions for this release. This section describes new features that were introduced in the initial 8.1 release of SIP Server.
Alert-Info
header into INVITE
requests, in order to specify a distinctive ring-tone
depending on the type of call.
Diversion
header for
redirected calls.
P-Early-Media
header, used to control the flow of media in
the early dialog state. P-Access-Network-Info
header, used to provide access to
network information about the user. REFER
to an outgoing INVITE
or REFER
. INVITE
or REFER
. SUBSCRIBE/NOTIFY
method for providing reliable MSML-based media
services.
OtherDN
handling. SIP Server now supports converting the Agent
ID to the corresponding DN in certain T-Library messages where the Agent
ID is included as the value of the OtherDN
field.
Dest-Capacity
key-value pair in the Extensions
Attribute, as applied by
the URS routing strategy, to set the capacity for a targeted Trunk.
Refer-To
header of a REFER message, by using the override-domain-oosp
configuration option.
SUSPENDING
or SUSPENDED
state before they are finally stopped. For
more information, refer to the Framework 8.1 SIP Server High-Availability Deployment Guide.
onunreach, unreach-timeout,
and onnotreg.
msml-support
parameter is enabled in the SIP Server application, SIP Server can now process the user data in MSML INFO
messages, propagated by GVP, and subsequently dispatching them to T-Library clients in corresponding T-Library events. (ER# 255608171)
00-05150
: Application's run mode changed to Primary. SCS generates this log event on behalf of any application when the application starts to run in Primary mode.
00-05151
: Application's run mode changed to Backup. SCS generates this log event on behalf of any application when the application starts to run in Backup mode.
Using log events generated by SCS removes the role of Message Server in a SIP Server switchover. (ER# 187865845)
geo-location
option when selecting a softswitch DN (Voice over IP Service
DN with service-type
set to softswitch
) located behind the switch. (ER# 275207979)
voicemail-pattern-x
DN-level option. This voicemail-pattern is used for integration with the Outbound IP Solution. It is used by SIP Server for matching the redirectNumber
header of a 181 Call Is Being Forwarded
message. For details, see the Framework 8.1 SIP Server Deployment Guide. (ER# 275279052)
Require: timer
header in hold INVITE
requests. (ER# 277838474)
From
header, a new configuration option, sip-pass-from-parameters
, is added.TServer
From
header and include them in to an outgoing INVITE
message, except a tag
parameter, which is generated by SIP Server. To pass all parameters from the From
header, set this option to ∗ (an asterisk). (ER# 279731621, 280033391)
This release includes the following corrections and modifications:
SIP Server no longer rejects a new incoming call that arrives at a DN if a previous call is released while SIP Server was processing a TMakeCall
request on behalf of that DN. This issue occurred if the refer-enabled
and reject-call-incall
configuration options were set to true
on the DN. (ER# 275680792)
SIP Server will now reschedule a re-INVITE
request when it receives a 503 Service Unavailable
message with the Retry-After
header in response to the initial re-INVITE
. This does not include cases where the 503 message is received for a new INVITE
. (ER# 231308403)
SIP Server can now play a specific ring-tone for the caller when an inbound call is routed to the agent's phone and the agent's phone starts ringing. To support this feature, you must configure the DN-level option sip-ring-tone-mode
to the value of 1
. Previously, SIP Server did not play ring-tones to the caller when the agent's phone was ringing, and the call was released after 32 seconds.
Note: To handle cases when the agent does not answer the call, you must configure SIP Server to clear stuck calls (requiring configuration at both the Application and DN-level). For more information, see the "No-Answer Supervision" section of the Framework 8.1 SIP Server Deployment Guide, which provides details about the parameters used for clearing NO ANSWER
calls. (ER# 221193433)
SIP Server now accepts TRegisterAddress
requests for any DN that is not configured in the configuration environment if the value of the ControlMode
attribute of the request is Local (2)
. Previously, SIP Server ignored this attribute and accepted registration for a non-configured DN only if its name was exr
. (ER# 241528277)
SIP Server now includes its own address in the Refer-To
header of REFER
messages that it sends to DNs behind a softswitch. Previously, when processing a TMakeCall
request for a DN behind a softswitch, SIP Server sent the REFER
to the DN that made the call using the Refer-To
header that contained the address of the softswitch. By using the SIP Server address instead, the process now works the same way as if the DN was configured with the contact
option. (ER# 235318785)
SIP Server now rejects THoldCall
and TRetrieveCall
requests received during a SIP session refresh. Previously, SIP Server accepted these requests during a session refresh, and they were not processed correctly. (ER# 250580130)
SIP Server now correctly fills the AttributeThirdPartyDN
field in EventReleased
messages sent during a 302 Moved
call forwarding scenario. (ER# 259742106)
SIP Server no longer mistakenly sends an INVITE
request for a music service to a particular party in cases where this party has initiated two consecutive INVITE
hold transactions. Previously, SIP Server sometimes sent this mistaken INVITE
. This situation only occurred when sip-enable-moh
was set to false
. (ER# 251513874)
SIP Server now applies Class of Service (COS) outbound dialing rules and ring-through rules configured at the Agent Login
level in deployments where SIP Server is located behind the softswitch. Previously, there options were applied only if COS was assigned to a Voice over IP Service
DNs with service-type
set to softswitch
. (ER# 258964162)
SIP Server no longer places calls on hold when a treatment is played through a Voice over IP Service
DN and the DN is configured with the charge-type
option set to 2
. (ER# 252245933)
SIP Server can now correctly process the Alternate Call operation (from a consultation call to a main call) if it receives a BYE
message from an external party during the incomplete re-INVITE
transaction. Previously, due to race conditions that occurred during the Alternate Call operation, the main call was stuck and SIP Sever was unable to retrieve the consultation call. (ER# 255822619)
SIP Server is now able to release a reserved agent DN so that the agent can make its next call. Previously, SIP Server did not release the reserved DN in cases where the REFER
for a consult call was rejected, leaving the DN unable to make a subsequent call. (ER# 256106372)
SIP Server now correctly updates the contact
option when it receives a REGISTER
request. Previously, SIP Server was unable to update the contact
option when the registration took place when the contact
option was already deleted from the Configuration Layer database. (ER# 263562810)
If a caller to a Voice-Treatment Port does not respond to a SIP re-INVITE
request, this will no longer result in the Voice-Treatment Port becoming stuck in the off-hook state. (ER# 264528310)
The SIP Server behavior now complies with RFC 3261. It now resends an ACK
message for each retransmitted 302 Moved Temporarily
message it receives. (ER# 250389491)
SIP Server now marks the treatment service as BackInService
upon receiving a 200 OK
message from Stream Manager when an ACK
message is delayed from a media gateway. Previously, in this scenario, SIP Server was not able to continue processing further T-Library requests on the call because of the late ACK
message. (ER# 256975329)
When a call is routed from SIP Server A to SIP Server B, SIP Server B is now able to use the ISCC pullback
transaction type to route the call back to SIP Server A. (ER# 260161347)
SIP Server now responds with a 500 Server Internal Error
message to an UPDATE
request that contains the SDP in pre-answered state. (ER# 261520143)
SIP Server now supports media service reliability for call recording services with the following limitation which is mentioned in the SIP Server 8.1 Deployment Guide:
Feature Limitation: SIP Server supports reliability for media services after the initial failure of a media server only. For any subsequent media server failure, SIP Server is unable to restart the service using another media server. (ER# 260298378)
SIP Server now supports an unlimited number of the dial-plan-rule-<n>
options. (ER# 259638066)
With the new logout-on-out-of-service
configuration option, SIP Server now controls whether it sends EventAgentLogout
when the agent device goes out of service. (ER# 262964635)
When the Class Of Service (COS) feature is configured at the Application level, it now applies only to DNs of type Extension
and ACD Position.
(ER# 261573390)
The SIP Server behavior has been corrected in the following scenario:
coach
mode.coach
mode.
During a multi-site supervision session, by using a TSetMuteOff
request, the supervisor's connection to the monitored call can now be changed between the initial MonitorMode
and an open supervisor presence. Previously, the observing party in this scenario was sometimes unable to hear the conversation (TSetMuteOff
request was not processed). (ER# 264558881)
SIP Server now synchronizes the DN state in High-Availability (HA) deployments when a TCallForwardCancel
request is issued by a T-Library client. (ER# 265150954)
SIP Server no longer releases the caller leg when it receives a BYE
message in response to a re-INVITE
message sent to Stream Manager. Previously, in this scenario, SIP Server incorrectly released the caller leg, resulting in a stuck call. (ER# 264724714)
SIP Server now sends an EventError
message before the EventReleased
if, during a THoldCall
operation, the DN that should be placed on hold does not answer the call. (ER# 264062329)
A new configuration option, sip-legacy-invite-retr-interval
, is added to support INVITE
retransmissions in accordance with RFC 3261 "SIP: Session Initiation Protocol." See the Framework 8.1 SIP Server Deployment Guide for details. (ER# 263419111)
When a recorded call that has been transferred to a monitored agent is released, SIP Server now releases all recording resources. Previously, these resources had to be released by Media Server (Stream Manager or Genesys Voice Platform) when the rtcp-inactivity-timeout
option was configured. (ER# 265759279)
The info-pass-through
option no longer has any effect on the collection of DTMF digits (it only controls whether SIP Server will proxy INFO
messages to its peer or not). Even if this option is set to true
, SIP Server will collect the digits when a treatment is in progress. (ER# 266763796)
When URS plays multiple PlayApplication
treatments, SIP Server now updates UserData
in between the processing of treatments. Previously, SIP Server did not send updated UserData
in the MSML dialog that was sent to start the second treatment. (ER# 273512539)
SIP Server now properly releases a call in a race-condition scenario that occurred during processing CANCEL
requests, as follows:
BYE
request before the agent answers it using 1pcc.CANCEL
request to the agent's phone.200 OK
message in response to an INVITE
message.200 OK
in response to the CANCEL
request.ACK
message in response to the 200 OK
containing the SDP, and then it sends the BYE
request to the agent's phone. (ER# 262833436)
The partition-id
for a particular predictive outbound call is now assigned based on the partition-id
setting on the Routing Point
or Trunk Group
DN that the call is made from. (ER# 266616315)
SIP Server now processes a second TRouteCall
request properly when, during processing of the first TRouteCall
request, the routing timeout expired before the ringback treatment could start. Previously, SIP Server did not sent an INVITE
request to the destination while processing the second TRouteCall
request. (ER# 266307622)
SIP Server now correctly applies values of theinfo-pass-through
configuration option, as specified in the SIP Server 8.1 Deployment Guide. (ER# 269277011)
The DN state is now correctly synchronized between primary and backup SIP Server instances after receiving a REGISTER
request containing the expires=0
parameter. Previously, the DN state was incorrectly changed to out-of-service on the backup SIP Server after receiving such a REGISTER
request. The issue occurred when the use-register-for-service-state
option was set to false
(the default value). (ER# 272090167)
SIP Server now correctly reports an EventDestinationBusy
message if an outbound call is made to a busy destination. (ER# 265057195)
SIP Server now retrieves a call using the re-INVITE
procedure if the call was earlier placed on hold with the re-INVITE
procedure as well. Previously, SIP Server sometimes sent a NOTIFY
message with the talk
event to the endpoint that was placed on hold by the re-INVITE
procedure. As a result, the retrieval was not successful.
Note: This issue occurred when the options dual-dialog-enabled=false
and sip-cti-control=talk,hold
were used simultaneously. Genesys does not recommend this type of configuration. (ER# 266258753)
SIP Server now provides high availability in a call-recording reliability-failover solution (NETANN and MSML). The new primary SIP Server is now able to recover the call recording if the recording device fails after a switchover. (ER# 265579769)
SIP Server now correctly processes the Replaces
parameter in the Refer-To
header. Previously, if angle brackets were missing in the URI scheme in the Refer-To
header, SIP Server did not process the Replaces
correctly. (ER# 268245231)
SIP Server now correctly changes its status to SERVICE_UNAVAILABLE
after the maximum number of attempts to listen to the port is reached (specified by the sip-max-retry-listen
option). When a port becomes available, SIP Server changes its status to RUNNING
. Previously, if SIP Server was finally able to open a SIP port, it remained in the SERVICE_UNAVAILABLE
status.
(ER# 266748364)
SIP Server now terminates subscription notifications from GVP (to get Resources info and Media Server status) after a SIP Server switchover. Previously, SIP Server did not terminate the subscription, which remained active until the subscription timer expired. (ER# 272893339)
In scenarios where SIP Server sends a REFER
request, SIP Server now processes the NOTIFY
message even when it is received before the 202 accepted
message sent in response to the REFER
. (ER# 274817923)
In a call-monitoring scenario where a supervisor rejects a call with a 480 Temporarily Unavailable
message, SIP Server now generates an EventReleased
message. Previously, in this scenario, SIP Server generated an EventAbandoned
instead, which negatively impacted reporting statistics. (ER# 277856987)
SIP Server now correctly applies the sip-ring-tone-mode
option setting for a softswitch DN (a DN of type Voice over IP Service
with server-type
set to softswitch
). Previously, the ringback was not played for a softswitch DN when sip-ring-tone-mode
was set to 0.
(ER# 277324490)
SIP Server now correctly maps T-Library UserData
containing local characters to a corresponding SIP header based on the userdata-map-filter
configuration option. (ER# 276064371)
If an SDP has been negotiated in a provisional request/response, SIP Server no longer sends a new SDP to the second party after receiving an empty 200
message from the first party. Instead, SIP Server sends the negotiated SDP. (ER# 270512766)
While processing a TApplyTreatment
request, SIP Server now correctly processes the error response that is received from Media Server in response to an INVITE
request, and also correctly responds to subsequent TApplyTreatment
requests from URS for the same call. Previously, in this scenario, the call sometimes became stuck. (ER# 278349528)
When an agent currently being monitored for voice is transferred to an emergency number, voice monitoring now stops correctly. (ER# 266227999)
For inbound calls only (where a dial-plan is associated with a Trunk
DN), SIP Server now forwards calls as specified by the dial-plan parameters ontimeout
, ondnd
and onbusy
included in the dial-plan rule. (ER# 263362013, 263392211)
SIP Server no longer becomes unstable in the following scenario:
sip-replaces-mode
is set to allow use of Replaces
on a device (setting 1
or 2
in the configuration option).INVITE
with Replaces
.INVITE
.
SIP Server now dynamically updates options in the INVITE
section. Previously, changes to the options did not take effect until after SIP Server was restarted. (ER# 257398165)
When operating with Alcatel-Lucent IMS, after a switchover, SIP Server now releases a call when an external party is released after the switchover. (ER# 249101697)
SIP Server now plays a second Music
or
Announcement
treatment on a Routing Point, if the
sip-early-dialog-mode
option on the Trunk
DN is set
to 1
and the ringing-on-route-point
option on the SIP
Server Application
object is set to true
.
(ER# 220926191)
SIP Server no longer reports a call as released if the re-INVITE
request to a call party results in the 5xx (Server Error)
response message and if the 500/503
is received with the Retry-After
header, then SIP Server will retry the hold re-INVITE
request later. In case of 1pcc hold, SIP Server forwards the error response received from one site to its counterpart. (ER# 169145901)
SIP Server now distributes an EventPartyAdded
message to
the conference controller (DN2) in the following scenario:
route
is used, and the main and
consultation calls are initiated via the same External Routing Point.(ER# 186552419)
SIP Server no longer mistakenly distributes a DNBackInService
event if
the properties of the corresponding DN are changed in the Configuration Layer.
(ER# 10324969)
SIP Server now supports single-step transfers of inbound calls to the PSTN in IP Multimedia Subsystem (IMS) deployments. (ER# 261457151)
In a scenario where routing to the destination with early media negotiation is in progress, SIP Server now correctly responds to the caller's hold INVITE with a 200 OK message containing a hold SDP. Previously, in this scenario, SIP Server responded with a 200 OK containing a non-hold SDP. This issue applied to scenarios where routing was initiated with a dummy SDP. (ER# 270877319)
This section provides corrections and updates for issues found in currently released documentation for this product. The changes described here will be included in future published versions of the document.
Agent Assist functionality is not enabled for the consult call even if the Consult Agent has an Agent Assist profile set up in options or Agent Assist parameters are sent via a RouteCall
request. This means, SIP Server will not send Agent Assist parameters in the INFO message to GVP, to start recording for a Consult Agent. Previously, Agent Assist parameters, if available for a Consult Agent, were sent by SIP Server to GVP when recording started for the consult call.
Related feature documentation: Passing Extended Recording Metadata to GVP in the Framework 8.1 SIP Server Deployment Guide.
(SIP-28199)
Because of the SIP-27886 issue correction in the 8.1.104.34 release, SIP Server no longer generates EventDNBackInService messages when some DN-level configuration options are added, updated, or deleted. That is why the following statement in the SIP Server Deployment Guide—Endpoint Service Monitoring chapter, Passive Out-of-Service Detection section—is no longer correct:
When the device entry in the Configuration Layer is changed. An EventDNBackInService message is generated in all scenarios. (SIP-27917)
This section provides the latest information on known issues and recommendations associated with this product.
Starting with release 8.1.103.65, SIP Server requires FlexNet Publisher License Manager version 11.13 or later. (SIPSVRDOC-596)
Found In: 8.1.103.65 | Fixed In: |
SIP Server does not propagate mapped key-value pairs with empty values from UserData to SIP headers of generated INVITE messages. (SIP-25115)
Found In: 8.1.103.25 | Fixed In: 8.1.103.37 |
SIP Server terminates unexpectedly while trying to reconnect to SIP Feature Server when one of the Feature Server nodes is down. (SIP-23463)
Found In: 8.1.102.95 | Fixed In: 8.1.102.98 |
If an error response is received from an active (running) Media Server (or the timeout expires, or there is no response to INFO) during a service recovery provided by this Media Server, it may cause recording do not clear correctly from the DN and new calls will be rejected for this DN. (SIP-20122)
Found In: 8.1.102.10 | Fixed In: 8.1.102.29 |
When an agent is logged in to the phone and the phone subscription expires, SIP Server does not log the agent out. (SIP-18246)
Found In: 8.1.101.56 | Fixed In: 8.1.101.57 |
In the early dialog state, if SIP Server receives an UPDATE message with the P-Asserted-Identity
header after the initial INVITE, SIP Server may not include the P-Asserted-Identity
header that was present in the UPDATE message while generating the subsequent INVITE for the destination. (SIP-17311)
Found In: 8.1.101.43 | Fixed In: 8.1.101.46 |
In multi-site scenarios, SIP Server rejects a TDeleteFromConference request with EventError whenever the request is submitted to remove the conference initiator, or the conferenced party present at the other site, if this party was not added by the party that submitted the request. (SIP-16797)
Found In: 8.1.101.30 | Fixed In: 8.1.101.42 |
SIP Server does not support an SDP change during a session refresh. If SIP Server receives the SDP change request during the session refresh, it does not propagate the SDP change to its peers. (SIP-15198)
Found In: 8.1.101.30 | Fixed In: |
In multi-site conference scenarios:
Found In: 8.1.101.25 | Fixed In: |
When OCS is registered on a Routing Point and is used in an outbound campaign, but the URS is not used in the same campaign for some reason (such as a URS failure or an incorrect configuration), the outbound call waits for the timeout specified by router-timeout
to expire before the call goes to the default DN instead of being routed immediately as specified by the default behavior. (SIP-16067)
Found In: 8.1.101.15 | Fixed In: |
If the sip-error-conversion
option is configured on both Application and DN levels, SIP Server only applies the Application-level value, ignoring the DN-level value, which should have a priority over the Application-level setting. (SIP-14967)
Found In: 8.1.100.99 | Fixed In: 8.1.101.04 |
SIP Server application configured to run in a single-thread mode
(sip-link-type=0
) does not release a call on a Hunt Group DN if
all Hunt Group members rejected this call and the default-dn
is
not configured for that Hunt Group DN or on the Application level. This call
is properly released if a caller hangs up. This call does not affect
distribution of other calls arriving to this Hunt Group DN. (SIP-14791)
Found In: 8.1.100.98 | Fixed In: |
For MSML-based recording, for multi-site calls with recording enabled at one site and the agent at another site, Genesys recommends that you disable video on MCPs from the resource group that provides the ''media'' service. (SIP-14829)
Found In: 8.1.100.93 | Fixed In: |
In the double-dip IMS environment where the server-role
configuration option is set to 3
and the
divert-on-ringing
option is set to false
, SIP Server
does not release the transferring party in the single-step transfer scenario
until the transfer destination answers the call.
Workaround: Genesys recommends setting the divert-on-ringing
option to true
at the Application level and to false
at the Routing Point DN where this functionality is required.
(SIP-14646)
Found In: 8.1.100.93 | Fixed In: 8.1.100.98 |
SIP endpoint's SIP registration via CUBE SBC (Registration pass-through) is not currently supported.
Found In: 8.1.100.74 | Fixed In: |
When the Active-Active RM pair is deployed in SIP Server using the IP-based
model (using the contact-list
option), there is the following
limitation related to high-availability (HA):
When msml service is established and switchover occurs, if the MCP used for the service goes down, RM notifies SIP Server that the service is down. SIP Server will not recover this service.
Workaround: Use the FQDN-based model (DNS SRV) of Active-Active RM pair deployment to avoid this issue.
(ER# 323837272)
Found In: 8.1.100.74 | Fixed In: |
For a deployment in which SIP Server is integrated with SIP Proxy, the value
of the Request-URI address specified in the request-uri
option
configured at the DN level must have the real host and port of the
corresponding DN. Any other values will cause incorrect results.
(ER# 321009728)
Found In: 8.1.100.74 | Fixed In: |
SIP Server does not support values 0
and max
for
the num-of-licenses
and num-sdn-licenses
configuration
options. You must explicitly specify the number of licenses of a certain type
allocated for a particular SIP Server. (ER# 319120151)
Found In: 8.1.100.64 | Fixed In: 8.1.100.74 |
When running on the Linux/UNIX operating system with TLS enabled, the TLS connection between SIP Server and its clients might be unstable, and SIP Server might terminate unexpectedly if the sip-tls-cipher-list
configuration option was specified. This issue affects SIP Server versions 8.1.100.64 and later. (SIP-16071)
Found In: 8.1.100.64 | Fixed In: 8.1.101.22 |
In a HA configuration, with the sip-iptakeover-monitoring
set
to true
, if the backup SIP Server detects that the IP address
specified by the sip-address
option is present on the backup
host, the Virtual IP address down control script is executed to remove that
IP address on the backup host. In that case, the Virtual IP address up control
script on the primary host is not executed. (ER# 316666221)
Found In: 8.1.100.64 | Fixed In: 8.1.100.74 |
If the dial plan fails, SIP Server plays a busy tone and releases a call.
For backward compatibility, if the dial plan is set to reject
,
for SIP Server to reject a call, the sip-busy-type
option must
be set to 2
. (ER# 317237570)
Found In: 8.1.100.64 | Fixed In: 8.1.100.74 |
When running on a Windows platform, SIP Server may experience memory growth after startup that should not exceed 15% of initial memory usage. This memory growth is due to expanding internal memory structures related to call and SIP registration processing, and the growth will eventually slow down and stop. (ER# 313092581)
Found In: 8.1.100.64 | Fixed In: |
If support for RFC 3263 is enabled, SIP Server experiences a memory leak of approximately 100 bytes for every non-INVITE SIP dialog that it initiates (when acting as a UAC).
Workaround: Upgrade to SIP Server release 8.1.1. This leak does not occur in release 8.1.1 or later.
(SIP-14138)
Found In: 8.1.001.29 | Fixed In: |
SIP Server does not add the mandatory cnonce
parameter in the Authorization
header if it receives the qop
directive in the WWW-Authenticate
header field. (ER# 312490411)
Found In: 8.1.001.10 | Fixed In: 8.1.100.94 |
SIP Server returns an EventError "Unknown cause"
message when the Authentication procedure fails. (ER# 312768969)
Found In: 8.1.001.10 | Fixed In: 8.1.100.98 |
When record-consult-calls
is enabled, dynamic call recording that is started for a consultation call will be stopped after the transfer is completed. An agent must explicitly initiate dynamic recording for merged calls even if record-consult-calls
and record-after-merge
are enabled. In an upcoming release, the original behavior will be restored, dynamic call recording that is started for a consultation call will be continued after the transfer is completed. (ER# 285146241)
Found In: 8.1.000.88 | Fixed In: |
If a DN subscribes to a message-summary
event, but the DN has no mailbox configured (option gvm_mailbox
), SIP Server accepts the subscription and sends an empty NOTIFY
message, but without the Message-Account
parameter. (ER# 299401974)
Found In: 8.1.000.76 | Fixed In: 8.1.100.64 |
SIP Server does not process TCompleteTransfer
requests while a consultation call is in the dialing state and the call is queued at a Routing Point on a remote SIP Server. (ER# 313065741)
Found In: 8.1.000.51 | Fixed In: |
SIP Server fails to apply the start_timeout
timer if the USER_ANN_ID
parameter is included in a PlayAnnouncementAndDigits
treatment request.
(ER# 290509215)
Found In: 8.1.000.49 | Fixed In: 8.1.100.64 |
A nailed-up session is not synchronized between primary and backup SIP
Servers. After a switchover, SIP Server sets up a new nailed-up connection
to a gcti::park
device. (ER# 285305941)
Found In: 8.1.000.37 | Fixed In: 8.1.100.91 |
If SIP Server is integrated with ALU IMS, the session-refresh-interval
configuration option must be set to a value of 100
sec or greater. If the timer value is less than 100
sec, ALU IMS drops the call. (ER# 263922681)
Found In: 8.1.000.37 | Fixed In: |
SIP Server does not unmute the party when a conference call is released. (ER# 279765616)
Found In: 8.1.000.37 | Fixed In: 8.1.000.93 |
When operating with the Siemens OpenScape Voice PBX, some issues may occur in the following scenario:
Found In: 8.1.000.37 | Fixed In: |
SIP Server does not send EventRouteUsed
with Cause
set to Overflow
and EventDiverted
with Cause
set to Redirected
in the following scenario:
Found In: 8.1.000.37 | Fixed In: 8.1.000.52 |
If an incoming INVITE
for an inbound call contains the SIP Alert-Info
header, and the Alert-Info
header is also specified in either the DN or Application-level options, then two Alert-Info
headers will be present in the INVITE
request that is sent to the endpoint. (ER# 280130309)
Found In: 8.1.000.37 | Fixed In:8.1.000.45 |
When operating in Disaster Recovery mode, in a scenario where a SIP registration expires on an agent DN that is configured to operate in dr-forward=no-agent
mode and is currently on the call, SIP Server logs the agent out, and does not send the expected EventDNOutOfService
message. (ER# 279059731)
Found In: 8.1.000.37 | Fixed In: 8.1.001.10 |
When an agent transfers a call to an external destination, SIP Server places itself in the Out Of Signaling Path (based on the option setting oosp-transfer-enabled=true
)and sends REFER to the caller even though the caller did not send the REFER support in the Allow header of the initial INVITE. (SIP-19825)
Found In: 8.1.000.37 | Fixed In: |
In 1pcc hold transactions, SIP Server does not pass custom headers in 200 OK
to a hold controller, in cases where the Application-level option sip-enable-moh
is set to false
and no media server is involved. (ER# 278355561)
Found In: 8.0.400.80 | Fixed In:8.1.000.52 |
In a scenario where a consultation call is placed on hold and the related INVITE
message is sent with the incorrect SDP, SIP Server may drop the call. (ER# 278671838)
Found In: 8.0.400.75 | Fixed In: |
TSCP issue that is applicable only to HA deployments. HA T-Server or SIP Server may become unstable in an environment where a new application object is created with a connection to the running HA T-Server or SIP Server. This can occur in any of the following scenarios:
Found In: 8.0.400.70 | Fixed In: 8.1.000.40 |
When an agent currently being monitored for voice is transferred to an emergency number, voice monitoring does not stop as it should. (ER# 266227999)
Found In: 8.0.400.45 | Fixed In: 8.1.000.37 |
For inbound calls only (where a dial-plan is associated with a Trunk
DN), SIP Server is unable to forward calls as specified by the dial-plan parameters ontimeout
, ondnd
and onbusy
included in the dial-plan rule. (ER# 263362013, 263392211)
Found In: 8.0.400.42 | Fixed In: 8.1.000.37 |
SIP Server does not support single-step transfers of inbound calls to the PSTN in IP Multimedia Subsystem (IMS) deployments. In these deployments, SIP Sever can complete single-step transfers of inbound calls to IMS users only. (ER# 261457151)
Found In: 8.0.400.32 | Fixed In:8.1.000.37 |
SIP Server may reject a TInitiateTransfer
request with EventError
if an agent submits this request while both listening to a busy tone and attempting to make a consultation call to a busy destination. This issue can be avoided if:
TInitiateTransfer
request; or TInitiateTransfer
request after waiting for five seconds.Found In: 8.0.400.28 | Fixed In: |
SIP Server may become unstable in the following scenario:
sip-replaces-mode
is set to allow use of Replaces
on a device (setting 1
or 2
in the configuration option).INVITE
with Replaces
.INVITE
.Found In: 8.0.400.25 | Fixed In: 8.1.000.37 |
SIP Server is unable to dynamically update options in the INVITE
section. Changes to the options will not take effect until after SIP Server is restarted. (ER# 257398165)
Found In: 8.0.400.25 | Fixed In: 8.1.000.37 |
When the SIP Server application exits, it is not possible to guarantee that the log message GCTI_APP_STOPPED
(GCTI-00-05063) will be delivered to the Solution Control Server (SCS).
If a reaction for SIP Server stoppage is required, only the following log events are guaranteed to be generated:
5091|STANDARD|GCTI_SCS_APP_PLANNED_STOP|Application stopped by Management Layer as planned
This message is produced by SCS on behalf of any application that is stopped according to a request. The request may be received from the Solution Control Interface (SCI), through SNMP, or initiated by an alarm reaction.
5064|STANDARD|GCTI_APP_TERMINATED|Application terminated due to internal condition
This message is produced by SCS on behalf of the application, in cases where the application stops without any request (manual stop or crash).
Found In: 8.0.300.34 | Fixed In: |
When operating with Alcatel-Lucent IMS, after a switchover, SIP Server may not release a call even if an external party is released after the switchover. (ER# 249101697)
Found In: 8.0.300.34 | Fixed In: 8.1.000.37 |
SIP Server may abandon a call if TSendDTMF
and TTreatmentCollectAndDigits
requests are processed on the same server for the same call. This may occur if these requests have the same number of digits for processing and that number is more than one and the treatment is interruptible. As a workaround, for this particular scenario, Genesys recommends that you do not use the interruptible treatment. If you must use such a treatment, allow 1-2 seconds delay in a strategy between the EventTreatmentEnd
message and TRouteCall
request. (ER# 249451012)
Found In: 8.0.200.45 | Fixed In: |
Predictive calls may be dropped due to an SDP negotiation failure in environments where the Paraxip media gateway is used along with GVP, and GVP is configured to support video. Indication of this problem is a 200 OK
message generated by the Paraxip media gateway, which contains an SDP body with two audio media parts. To avoid this problem, disable video support on GVP.
(ER# 226801731)
Found In: 8.0.001.00 | Fixed In: |
SIP Server may be unable to send a CANCEL
request to the correct destination after an HA switchover, depending on DN configuration, in cases where the switchover occurs while the call is in a ringing state. RFC 3261 mandates that CANCEL
requests must be sent to the same destination where the originating INVITE
request was sent. This requirement might not be met if the destination DN is configured with the option request-uri
, where the value of this option does not match the URI of the INVITE
destination. In this case, SIP Server sends the CANCEL
to the destination specified by the request-uri
, instead of to the INVITE
destination as required by RFC 3261. (ER# 229358797)
Found In:8.0.000.12 | Fixed In: |
SIP Server may be unable to play a second Music
or
Announcement
treatment on a Routing Point, if the
sip-early-dialog-mode
option on the Trunk
DN is set
to 1
and the ringing-on-route-point
option on the SIP
Server Application
object is set to true
.
(ER# 220926191)
Found In: 8.0.000.12 | Fixed In: 8.1.000.37 |
Race conditions that lead to an incorrect audio path may occur in these scenarios:
direct-uui
.Extensions
attribute of the
TRouteCall
request.DN
or Agent
Login
objects.As a workaround, Genesys recommends that you configure a greeting only in one
place—for example, in the TRouteCall
request.
(ER# 212618872)
Found In:7.6.000.61 | Fixed In: |
A greeting does not work when the ISCC transaction type
direct-notoken
is used. (ER# 212618863)
Found In:7.6.000.61 | Fixed In: |
SIP Server reports a call as released if the re-INVITE
request
to a call party results in the 5xx (Server Error)
response message.
(ER# 169145901)
Found In: 7.6.000.40 | Fixed In: 8.1.000.37 |
SIP Server cannot process a conference back to GVP (Genesys Voice Platform)
when the request-uri
and From
headers contain the
same DNs. (ER# 177931740)
Found In: 7.6.000.40 | Fixed In: |
When an agent places a call on hold, Asterisk may report the agent presence status incorrectly. For more information, see your Asterisk documentation. (ER# 180100206)
Found In: 7.6.000.40 | Fixed In: |
SIP Server does not distribute an EventOutOfService
message if
a SIP endpoint is unplugged and the softswitch responds with a 606 (Not
Acceptable)
message to the INVITE
message during creation
of a new call. This issue is applicable to SIP Server that is integrated with
BroadSoft version 13. (ER# 181825291)
Found In: 7.6.000.40 | Fixed In: |
SIP Server may remove an observer from a monitored call that has the
following parameters: MonitorScope
is set to agent
and MonitorMode
is set to connect
. (ER# 183356827)
Found In: 7.6.000.40 | Fixed In: |
SIP Server does not invite a supervisor for a supervision session when the previous supervision attempt fails because of the MCU (Multipoint Conference Unit) malfunction. (ER# 185732135)
Found In: 7.6.000.40 | Fixed In: |
SIP Server does not attach more than 16 KB of user data to the call from a
SIP message even if the SIP Server's configuration option
user-data-limit
allows attaching more than 16 KB of the data.
(ER# 186526930)
Found In:7.6.000.40 | Fixed In: |
SIP Server does not distribute an EventPartyAdded
message to
the conference controller (DN2) in the following scenario:
route
is used, and the main and
consultation calls are initiated via the same External Routing Point.(ER# 186552419)
Found In: 7.6.000.40 | Fixed In: 8.1.000.37 |
SIP Server distributes a UserEvent
message that contains RTP
information to any registered DN, even if the DN registered without a password.
(ER# 96136766)
Found In: 7.5.000.15 | Fixed In: |
A RouteCall
request that contains the
RouteTypeReject
parameter does not terminate a chat dialog.
(ER# 114530456)
Found In: 7.5.000.15 | Fixed In: |
If an attempt to update the SIP registration information for an endpoint
with Configuration Server is unsuccessful, the contact
info in the
DN object will not be updated until the next SIP registration attempt.
(ER# 98944416)
Found In: 7.5.000.15 | Fixed In: |
The EyeBeam endpoint does not retrieve a call after the call was in
Hold
status because the INT-IP media gateway will not accept an
empty INVITE
request. (ER# 65460140)
Found In: 7.2.100.35 | Fixed In: |
SIP Server incorrectly updates the contact
option in the DN
configuration if the authentication process for the REGISTER
command fails. (ER# 49192671)
Found In: 7.2.001.27 | Fixed In: |
SIP Server allows a user to set the Do-Not-Disturb
feature when
a DN is in an out-of-service
state. (ER# 30340720)
Found In: 7.2.001.18 | Fixed In: |
SIP Server mistakenly distributes a DNBackInService
event if
the properties of the corresponding DN are changed in the Configuration Layer.
(ER# 10324969)
Found In: 7.1.001.09 | Fixed In: 8.1.000.37 |
This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list. For more information on discontinued support for operating environments and databases, see Discontinued Support in the Genesys Supported Operating Environment Reference.
The following operating systems:
Discontinued As Of:February 1, 2021; 8.1.104.07 |
Windows Server 2008 operating system
Discontinued As Of:8.1.103.98 |
Microsoft Visual Studio 2008 (Visual C++ 9) and prior versions
Discontinued As Of:8.1.103.65 |
Red Hat Enterprise Linux 5 operating system
Discontinued As Of:8.1.103.52 |
AIX 5.3 operating system
Discontinued As Of:8.1.103.06 |
Solaris 32/64-bit version 9 operating system
Discontinued As Of:8.1.103.06 |
Windows Server 2003 operating system
Discontinued As Of:8.1.102.95 |
Red Hat Enterprise Linux AS 4 operating system
Discontinued As Of:8.1.102.41 |
Solaris 8 operating system
Discontinued As Of:8.1.000.37 |
Chat functionality using Microsoft LiveCommunications Server (LCS)
Discontinued As Of:8.1.x |
Information in this section is included for international customers.
There are no known internationalization issues for this product.
Additional information on Genesys Cloud Services, Inc. is available on our Customer Care website. The following documentation also contains information about this software.
Framework 8.1 SIP Server Deployment Guide contains detailed reference information for the Genesys Framework 8.1 SIP Server, including configuration options and specific functionality.
Framework 8.1 SIP Server High-Availability Deployment Guide contains reference information related to SIP Server high-availability deployment options, workflows, and deployment procedures for each supported operating system.
Framework 8.1 SIP Server Integration Reference contains reference information related to integrating SIP Server with SIP softswitches and gateways.
Management Framework Deployment Guide helps you configure, install, start, and stop Framework components.
Genesys Events and Models Reference Manual contains the T-Library API, information on TEvents, and an extensive collection of call models.
Genesys Migration Guide contains a documented migration strategy for each software release. Please refer to the applicable portion of this guide or contact Genesys Customer Care for additional information.
The SIP Server page in the Genesys Supported Operating Environment Reference provides detailed information about the supported operating environments, including requirements, supported versions, and any conditions or limitations for SIP Server.
Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD.
Note: For the DVD, the New Documents on this DVD page indicates the production date for that disc. Due to disc production schedules, documentation on the Genesys Documentation website may be more up-to-date than what is available on disc immediately after a product is released or updated.