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is no longer an affiliate of Alcatel-Lucent; any indication of such affiliation within Genesys products or packaging is no longer applicable.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 8.0 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 Read Me. 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 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release contains the following corrections or modifications:
When working in multi-threaded mode, SIP Server no longer terminates unexpectedly while processing internal inter-thread messages in a race condition. (SIP-15635)
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 the angle brackets (< >) when passing the INVITE. Previously, SIP Server removed the angle brackets from the From header.
(SIP-15748)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release contains the following corrections or modifications:
Enabling network debug logging by setting the x-conn-debug-all configuration option to 1 no longer may cause SIP Server to terminate unexpectedly. (SIP-15074)
SIP Server now correctly handles UDP SIP traffic after receiving ICMP TTL Expired packets. Previously, after receiving such packets, SIP Server was not able to receive and process any messages received via the UDP protocol, which sometimes caused a SIP Server switchover requiring restart of the failed instance of SIP Server. This issue was reintroduced starting with version 8.0.401.05, because SIP Server was incorrectly built with the previous version of the connection library. Now this issue is fixed. (ER# 300134706)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release contains the following corrections or modifications:
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-13592)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 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 8.0 and/or 8.1 versions of SIP Server.
This release contains the following corrections or modifications:
If an agent with a nailed-up connection involved in main and consultation calls, which are placed on hold, and then 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 to the main call, which remains on hold. Previously in this scenario, the agent was not re-invited to the main call. (ER# 323107332)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release contains the following corrections or modifications:
After a switchover, SIP Server HA instances now correctly synchronize the SessionID attribute. Previously, the primary SIP Server created a new SessionID attribute while restoring the connection and, as result, T-Library clients were not able to reconnect to the backup SIP Server. (ER# 313769575)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release contains the following corrections or modifications:
SIP Server now correctly encodes the User Data and includes it in the encoded form in the SIP headers. Previously, SIP Server did not encode some of the German characters which resulted in incorrect TLib-to-SIP data mapping. (ER# 310778129)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 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# 305759053)
This release includes no 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 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release does not contain any 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 could result in SIP Server incorrectly placing certain DNs out of service due to mis-timed synchronization requests. (ER# 304077531)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
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# 300463971)
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# 301197851)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release does not contain any 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# 300134706)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly clears the reason code that is issued when an agent logs out. Previously, when this agent logged back in using the AgentManualIn mode, the previous reason code was sometimes incorrectly reflected in the EventAgentNotReady message.(ER# 295215283)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 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 processes the confirmed state in BroadSoft Notifications, in cases where the content of the NOTIFY message contains only one dialog with local and remote tag. Previously in this scenario, SIP Server unexpectedly generated an EventAgentReady message. (ER# 286134749)
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 message.INFO message and did not end the treatment. (ER# 288667739)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 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:
For Outbound Solution with Media Server using MSML, the primary SIP Server now correctly propagates call merge results (TMergeCalls) to the backup SIP Server. Previously, SIP Server did not propagate call merge results properly, which led to a memory leak in the backup SIP Server. (ER# 282573843)
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# 282467914)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.17. 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 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 version of SIP Server is built with TSCP 8.0.101.17, 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:
SIP Server now properly generates voice monitoring events (EventUserEvent) that indicate the start and the end of the conversation of the 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# 281575394, 280116792)
SIP Server now correctly releases the transfer destination party if a caller disconnects the call while the 3pcc CompleteTransfer operation is in progress and the consultation call recording is enabled. (ER# 281991919)
SIP Server no longer becomes unstable if the MSML treatment timeout occurs after the TRouteCall request is issued.
(ER# 281417011)
SIP Server now correctly applies the Active Out-of-Service detection mechanism to DNs of type Trunk Group.
(ER# 282346302)
SIP Server now consistently generates an EventTreatmentEnd message when a race condition occurs during execution of the PlayAnnoucementAndDigits treatment. This race condition may occur if SIP Server does not receive DTMF input within the TIMEOUT interval specified in the treatment, but then receives the DTMF tone value in a SIP INFO message from Media Server just before the TIMEOUT interval elapses. (ER# 276766852)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.15. 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 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 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# 279428674)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.15. 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 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:
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 or TRouteCall requests from URS for the same call. Previously in this scenario, the call sometimes became stuck. (ER# 280991933, 278962969)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.15. 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 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 DNs are removed and re-created in the configuration environment, they will no longer have an incorrect contact information. Previously, SIP Server could sometimes use an incorrect DN contact information after such changes were made in the configuration environment. Note that the DN contact information obtained through the SIP REGISTER procedure takes precedence over the contact changes made in the configuration environment. (ER# 279561857)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.15. 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 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:
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
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# 277670991)
SIP Server now correctly unmutes the party when a conference call is released, as in accordance with the TSetMuteOff request for established conferences. (ER# 278184953)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.15. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature or functionality.
TServer section: 00–65535REFER method is completed.This release includes the following corrections and modifications:
This release of SIP Server is built with TSCP 8.0.101.15, which corrects the following issue:
In rare scenarios, involving dynamic update of Host objects in the configuration environment, SIP Server could become unstable. (ER# 276929084)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly passes custom headers (from the INVITE) between the dialogs during re-INVITE transactions initiated by the called party, in cases where the Application-level option sip-enable-moh is set to false. Previously in this scenario, SIP Server was sometimes unable to send custom headers. (ER# 276475329, 278031204)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
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# 271381967)
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, and so, after the switchover, the agent remained in the Ready state when the call was released. (ER# 269507484)
SIP Server now correctly processes the following scenario:
TReconnectCall request.
Previously in this scenario, the call became stuck. This issue occurred if the dual-dialog-enabled option was set to false. (ER# 272858872)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
When a Media Server goes down after the call treatment is established, SIP Server now clears the existing call treatment with that Media Server, and reestablishes it with a different Media Server. The msml-support configuration option must be set to true. (ER# 275812421)
SIP Server no longer has memory leaks that previously occurred in call-routing scenarios. (ER# 275754375)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
When operating with GVP Media Control Platform (MCP)/Genesys Media Server, SIP Server now properly handles race conditions that may occur in the following scenario:
msml-support option is set to true.EventTreatmentApplied to URS. URS requests to play another announcement B.dialogend to the previous announcement A, starts a dialog to announcement B, and sends to MCP the INFO message using MSML.INFO message containing dialog.exit to SIP Server.INFO received from MCP and sends a dummy 200 OK response to MCP.200 OK message from MCP in response to the INFO message sent for playing announcement B.200 OK and sends EventTreatmentApplied to URS for announcement B.dialogend.silentfail option must be set to true in the msml section of the MCP application. Otherwise, if MCP responds with an error message, SIP Server will attempt to restart the treatment with another available Media Server. (ER# 274635639)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
In accordance with RFC 4028, SIP Server now does not initiate a session refresh if it will be initiated by its client. And, if a client is the one who should do the refreshing (contains a refresher parameter), but the session refresh is not initiated, SIP Server will drop the call leg related to that client after a certain timeout. Previously, SIP Server performed a session refresh for all call legs that could result in race conditions between session refresh requests from SIP Server and its client. (ER# 272268014)
SIP Server now properly cleans up calls that may become stuck after receiving a 404 Not Found message. (ER# 268139585)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.14. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
SIP Server can now cancel outbound Predictive Calls when the outbound gateway returns a voicemail number in the redirectNumber of the 181 Call Is Being Forwarded response to the initial outbound INVITE request.
The redirectNumber header includes the number to which the call is being redirected, as well as the reason for the redirection. If SIP Server matches the number in the header to the configured voicemail pattern in the outbound Trunk configuration, SIP Server cancels the call, mapping the reason to a Genesys CallState attribute (for example, Busy).
To support this functionality, a new DN-level configuration option, voicemail-pattern-<n>, has been introduced. You must configure this option in the Trunk DN representing the outbound calling gateway. For example, in integrations with HP OpenCall Media Platform (OCMP), configure this option in the Trunk DN that represents the OMCP.
voicemail-pattern-<n>
Default Value: No default value
Valid Values: A string pattern in the Asterisk format
Changes Take Effect: At the next established call
Specifies the pattern that SIP Server looks for in the redirectNumber header of 181 Call Is Being Forwarded messages. If SIP Server matches the pattern in the header to the pattern configured in this option, it cancels the call, mapping the reason for the redirection to a Genesys CallState If SIP Server does not make a match between the header and this option, the call proceeds as normal.
The redirectNumber header arrives in the format
XYYYYYYYYYYYY
where X gives the reason for the redirection, and YYYYYYYYYYYY gives the number where the call is being redirected. SIP Server matches X to the corresponding Genesys CallState as follows:
| Code | Reason | Genesys CallState |
|---|---|---|
X=1 |
Call forward; line busy | Busy |
X=2 |
Call forward; no reply | NoAnswer |
X=3 |
Call forward; unconditional | AnsweringMachineDetected |
X=4 |
Call forward; not reachable/Other | AnsweringMachineDetected |
SIP Server uses the Asterisk pattern-matching format to match the value of the header to the value of this option. For a full description of Asterisk pattern-matching syntax, see the description for the Dial Plan feature in the Framework 8.0 SIP Server Deployment Guide.
You can also use the key-value pair voicemail-pattern in the Extensions Attribute of TMakePredictiveCall requests to match the redirectHeader to a particular voicemail number. Multiple patterns can be configured in a comma-separated list. If present in the TMakePredictiveCall, this value takes precedence over the Trunk configuration. (ER# 272695551)
This release includes the following corrections and modifications:
If the connection to Configuration Server is lost and then restored, SIP Server now correctly re-connects to Configuration Server and re-reads the configuration data. (ER# 267499690)
This version of SIP Server is built with TSCP 8.0.101.14, which corrects the following issue:
TSCP internal objects have been redesigned to prevent an exception that may occur during termination of the SIP Server process. (ER# 271155430, 268558501)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 SIP Server.
This is a hot fix for this product. This release contains no new features or functionality.
This release includes the following correction and modification:
In Out of Signaling Path (OOSP) scenarios (oosp-transfer-enabled=true ), SIP Server now places the URI content in angle brackets (< >) for the Contact header in 302 Moved messages and for the Refer-To header in REFER messages. (ER# 272003732)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
SIP Server now supports a configurable limit for the length of data included in the User-to-User Information (UUI) header, up to a maximum of 8192 characters. To support this functionality, a new Application-level option has been introduced.
sip-max-uui-length
Default Value: 256
Valid Value: 0-8192
Changes Take Effect: Immediately
Specifies the maximum number of characters by which SIP Server limits the length of the data included in the User-to-User Information (UUI) header. For example, with the default value of 256, SIP Server will limit the length of UUI content, encoded with hexadecimal encoding, to 128 bytes. (ER# 271241360)
This release includes the following corrections and modifications:
SIP Server now correctly forwards NOTIFY requests with message-waiting-indication (MWI) through to the destination endpoint. Previously, SIP Server could accept the NOTIFY with MWI, but not forward it to the endpoint. (ER# 271360504)
When SIP Server parses a URI field in the Contact header of the message, it no longer returns a 416 Unsupported URI Scheme message if some URI field parameters are in upper case. This SIP Server behavior now complies with RFC 3261, which states that when comparing header fields, field names are always case-insensitive. (ER# 270693488)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Support for the 911 Enable (E911) Emergency Gateway (EGW) versions 3.2 and 3.3. SIP Server now supports 911 emergency calling using the integration with the E911 EGW and service. When properly configured, 911 calls made from devices registered to SIP Server are sent through the EGW. Emergency calls can be made directly from the phone (1pcc call) or by request from a T-Library client (3pcc call); if both methods are available, 1pcc rather than 3pcc is recommended. As well, the EGW integration allows the Public Safety Answering Point (PSAP, or 911 dispatch) to discover the location of the dialing device and to provide a Call Back Number (CBN), in case the call is prematurely disconnected.
The following table lists the key actions you must complete to integrate SIP Server with the E911 Emergency Gateway.
| Step | Action | ||||||||
|---|---|---|---|---|---|---|---|---|---|
|
SIP Server uses this DN to place the outgoing 911 call as initiated by the SIP endpoint.
|
||||||||
|
For environments that allow DID calls, do the following (click to expand):
| ||||||||
|
For environments that do not allow DID calls, do the following (click to expand):
| ||||||||
|
|
||||||||
|
Genesys provides a utility that extracts a list of
|
To support this feature, new DN-level options emergency-backup, emergency-device, and emergency-callback-plan have been added to this release of SIP Server. You can configure these options on DNs of type Trunk Group or Trunk.
Note: For the callback Trunk, this option must contain the single address of the backup EGW only.
emergency-callback-planSpecifies the name of the Voice over IP Service dial-plan DN created for integration with the 911 Enable (E911) Emergency Gateway (EGW). For deployments that support Direct Inward Dialing (DID), you must:
Configure the dial plan itself so that its dialing rule converts the calling DN (ANI) into a 10-digit call back number (CBN) that SIP Server will include in the P-Asserted-Identity header of INVITE requests it sends on behalf of registered DNs, when processing 911 calls.
Trunk Group DN representing the EGW, set the emergency-callback-plan option to the name of the ANI-to-CBN dial-plan DN.Note: For deployments that do not support DID, do not configure this option. Instead, you must configure the Trunk DNs to represent the EGW as described in Step 3 of the previous table.
falsetrue, falseIf set to true, enables this device to conduct emergency calls.
SIP Server now supports the following log events that are generated by Solution Control Server (SCS) when SIP Server changes run modes, as follows:
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# 274659245)
This release includes the following corrections and modifications:
SIP Server now properly handles a race-condition scenario and connects the call with the answered party if the no-answer timeout expires while SIP Server was processing a TAnswerCall request. Previously in this scenario, SIP Server incorrectly released the call (or redirected the call to a no-answer-overflow DN) when the no-answer timeout expired. (ER# 269762767)
When processing an Out Of Signaling Path (OOSP) transfer scenario with the specified override-domain configuration option, SIP Server now correctly passes through User-to-User information in the Refer-To header of the REFER message. (ER# 270147341)
In scenarios where a TReleaseCall request is made in a dialing state, SIP Server now sends a non-zero audio port (m=audio xxxx) in the 200 OK message SDP. Previously in this scenario, SIP Server sent an audio port set to zero (m=audio 0) and the call was not released properly. (ER# 268298171)
When a conference initiator with recording enabled (record=true) places a conference on hold, SIP Server no longer mistakenly sends an EventRetrieved message. (ER# 268759335)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 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 re-INVITE message containing an SDP offer with the sendonly attribute during a pending REFER transaction by responding with a 200 OK message containing an SDP answer with the recvonly attribute. Previously, SIP Server responded with a 200 OK message containing the last-known SDP answer. (ER# 264886265)
In a 1pcc single-step transfer scenario, SIP Server now, on receiving a 180 Ringing or 200 OK message from a transfer destination, sends a NOTIFY request to the transfer-initiating party and waits for a 200 OK before sending the BYE message to that party. If SIP Server does not receive 200 OK within two seconds, it will send the BYE message to the transfer-initiating party and continue processing of 200 OK from the transfer destination. Previously in this scenario, SIP Server sent a NOTIFY followed by the BYE to the REFER leg of the call. (ER# 269094224)
SIP Server now correctly does not pass INFO messages according to the value "-" configured for the info-pass-through option. Previously, SIP Server did not apply option the value "-" correctly. (ER# 269299735)
SIP Server now correctly handles scenarios where the call destination sends multiple reliable provisional responses. Previously in this scenario, SIP Server responded with PRACK only to the first provisional response received from the call destination, but did not send PRACK to acknowledge the subsequent responses. As a result, the terminating party dropped incorrectly from the call. (ER# 262833661)
If Call Progress Detection is required in the Outbound Solution scenario, SIP Server now correctly applies the trunk selection algorithm by using the priority, capacity, or last-call time difference while choosing a trunk device. Previously, SIP Server did not apply the trunk selection algorithm correctly. (ER# 268236094)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
SIP Server now supports passing the P-Early-Media header, as per RFC 5009 "Private Header (P-Header) Extension to
the Session Initiation Protocol (SIP) for Authorization of Early Media." This header can be used to control the flow of media in the early dialog state. SIP Server supports the P-Early-Media header in the following messages: 18X, 200, INVITE, PRACK, and UPDATE.
The P-Early-Media header is passed only when both the calling and destination domains are configured with enforce-trusted set to true.
This functionality is only applicable if the calling side supports early media dialogs.
(ER# 267611092)
SIP Server now supports passing the P-Access-Network-Info header, as per RFC 3455 "Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)." This header can be used to provide access network, location, and emergency call information about the user agent. SIP Server supports the P-Access-Network-Info header only in 18X, INVITE, and UPDATE messages.
The P-Access-Network header is passed only when both the calling and destination domains are configured with enforce-trusted set to true.
This release includes the following corrections and modifications:
SIP Server now correctly sends a 200 OK message that contains the SDP with the hold status (a=inactive/sendonly) if the call is placed on hold when the destination does not answer. Previously in this scenario, SIP Server sent a 200 OK message without including the SDP with the hold status. (ER# 260627235)
If SIP Server receives an UPDATE message that contains the SDP, it now correctly includes the SDP when it generates a 200 OK response. Previously in this scenario, SIP Server sent a 200 OK message without including the SDP. (ER# 265799333)
In a scenario where SIP Server sends a dummy SDP in the INVITE message to the destination and at the same time receives the re-INVITE message with the hold SDP from the originating party, SIP Server now handles the re-INVITE properly and connects the originating party with the destination.
(ER# 266009891)
If an early dialog is terminated by a BYE request, SIP Server now correctly sends a 487 Request Terminated message to the INVITE originator. (ER# 266565282)
SIP Server now complies with RFC 3264 "An Offer/Answer Model with Session Description Protocol (SDP)," which states that the media type in an answer must match the media type specified in the offer. For example, if the caller puts a call on hold (INVITE with SDP a=inactive), SIP Server now replies with a 200 OK message that includes the hold status (SDP a=inactive).
To support this SDP-matching for sendonly, a new value passthrough has been added to the DN-level option sip-hold-rfc3264. This new setting prevents SIP Server from changing sendonly and recvonly SDP attributes to inactive in the answering SDP; in other words, SIP Server simply passes these attributes through unchanged. (ER# 267030271)
This version of SIP Server corrects a memory leak that may occur on the AIX platform in SIP Server version 8.0.300.39 and later. (ER# 267718327)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.09. 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 SIP Server.
This is a hot fix for this product. This release contains no new features.
This release includes the following corrections and modifications:
This version of SIP Server is built with TSCP version 8.0.101.09, which corrects the following issue:
When the resource-allocation-mode option is set to circular, SIP Server now correctly allocates resources in a circular manner within each epn partition (known to SIP Server by the partition name provided in the epn option in the Annex tab of any access resource DN). Previously, SIP Server allocated resources in a circular manner using a combined list of resources belonging to all partitions, which could result in a biased selection within any particular partition. (ER# 264905870)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features.
SIP Server can now pass custom SIP headers from a REFER request to an outgoing INVITE or REFER request. A new Application-level configuration option, sip-pass-refer-headers, must be configured to enable this functionality.
sip-pass-refer-headers
Default Value: An empty string
Valid Values: A string of SIP headers separated by commas; may contain full header names or name parts with an asterisk, representing a subset of headers
Changes Take Effect: At the next established call
When specified, SIP Server will pass custom SIP headers from a REFER request to an outgoing INVITE or REFER request.
For example:
If the sip-pass-refer-headers option is set to X-Tellme-*,X-Information and the incoming REFER request contains the following headers:
X-Tellme-Session-ID
X-Tellme-Header
X-Information
Then, SIP Server will include all three headers to an outgoing INVITE or REFER request.
(ER# 265839789)
SIP Server can now filter out Genesys internal SIP headers from TRouteCall and TMakePredictiveCall requests when generating an outgoing INVITE or REFER request to a media gateway. A new DN-level configuration option, enable-extension-headers, must be configured to enable this functionality.
enable-extension-headers
Default Value: predictive,routing
Valid Values: See value descriptions below
Changes Take Effect: At the next established call
This option controls which SIP headers, specified as a value in SIP_HEADERS of AttributeExtensions in TRouteCall and/or TMakePredictiveCall requests, are blocked or mapped from these T-Library requests into an outgoing INVITE or REFER request, as follows:
predictive,routing—Enables mapping of SIP headers from both TRouteCall and TMakePredictiveCall requests (the default behavior);none—Blocks mapping of SIP headers from both TRouteCall and TMakePredictiveCall requests;predictive—Enables mapping of SIP headers from TMakePredictiveCall requests only;routing—Enables mapping of SIP headers from TRouteCall requests only.enable-extension-headers option is set to routing on a Trunk DN and the TMakePredictiveCall request contains the following values in SIP_HEADERS of AttributeExtensions: AttributeExtensions
�HEADER1� �data1�
�HEADER2� �data2�
�HEADER3� �data3�
�SIP_HEADERS� �HEADER1,HEADER2,HEADER3�HEADER1, HEADER2, HEADER3 when generating an outgoing INVITE or REFER request to an external gateway.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 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature.
SIP Server now supports the SIP User to User Information (UUI) header, as specified in the RFC draft "A Mechanism for Transporting User to User Call Control Information in SIP". (ER# 266436441)
This release includes the following corrections and modifications.
SIP Server now correctly responds to 491 Request Pending messages that it receives after sending a re-INVITE request with hold SDP. Previously, SIP Server could drop the call on receiving the 491 Request Pending response. (ER# 262426790)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 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 multiple P-Asserted-Identity headers into an INVITE request. Previously, SIP Server could inadvertently insert more than one P-Asserted-Identity header. (ER# 266748265, 266786582)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T‑Server Common Part (TSCP) release number 8.0.101.05. TSCP is the shared software that all T‑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 any new features or functionality.
This release includes the following corrections and modifications:
SIP Server is now able to select an alternate destination in cases where a TMakePredictiveCall request fails by timeout or 503 Service Unavailable error. Previously, SIP Server released the call instead of retrying the operation on a different valid trunk. (ER# 264406853, 260321550)
SIP Server is now able to pass SIP INFO messages with Content-Type application/media_control+xml from Genesys Media Server to external parties in single-step conference scenarios. Previously, SIP Server was unable to pass these INFO messages even when configured to do so (the option info-pass-through is set to *). (ER# 266557161)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer becomes unstable while processing a predictive call if the INFO message with a CPD result from the Media Server is delayed beyond a specified timeout. (ER# 265579747)
SIP Server no longer becomes unstable while attempting to process registration requests from two T-Library–based clients having the same non-zero value in the SessionID attribute. (ER# 265791475)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly processes the ISCC transaction type reroute if an ISCCRequestGetCallDataXferService request from another SIP Server comes before a previous transaction is completed. (ER# 260769694)
SIP Server now correctly applies a replace-prefix option in call forwarding scenarios. Previously, SIP Server sometimes mistakenly applied the replace-prefix algorithm twice, causing incorrect trunk matching. (ER# 261663667)
After a switchover, SIP Server can now correctly transfer calls using the REFER method. Previously, after a switchover, SIP Server was not able to match the Trunk DN with the external calling device, which resulted in failed call transfer transactions. This issue occurred when the Trunk DN did not have a port value specified in the contact option. (ER# 263913042)
SIP Server now correctly handles two subsequent REGISTER requests containing expires=0 in the Contact header that are received from the same DN but in different dialogs. Previously in this scenario, SIP Server did not set the Expires header to 0 and did not reset the timer when a second REGISTER request containing expires=0 in the Contact header was received. (ER# 264389918)
SIP Server now correctly sends a 200 OK in response to a CANCEL request that is sent for a re-INVITE transaction, in accordance with RFC 3261. Previously, SIP Server did not send a 200 OK, which resulted in a stuck transaction. (ER# 264413648)
SIP Server now correctly processes user information from the P-Preferred-Identity header during the authentication procedure. Previously, only requests containing the P-Asserted-Identity header were processed correctly. (ER# 262253274)
After a switchover, SIP Server now reports agent states correctly. Previously, if a switchover happened right after the agent logged in, SIP Server sometimes did not report the agent state correctly. (ER# 264246017)
SIP Server now correctly routes a call to a newly created Routing Point. Previously, SIP Server sent a SIP/2.0 404 Not Found message in response to an INVITE message to such a Routing Point. This issue only occurred in deployments where SIP Server was obtaining its configuration from a Configuration Server Proxy HA pair. (ER# 265057124)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature.
Trunk DNs. SIP Server now supports assigning dial plans from Trunk DNs for inbound calls. Some limitations apply.This release includes the following corrections and modifications:
SIP Server now inserts the proprietary X-Genesys-Orig header only in call flows that require it. For example, in IMS deployments. This reduces unnecessary UserData processing. (ER# 262247568)
SIP Server no longer incorrectly ends the call on receiving an UPDATE message after an INVITE request for a TApplyTreatment request. Previously in this scenario, upon receiving the UPDATE, SIP Server did not apply a treatment and dropped the call. (ER# 263239377)
SIP Server now correctly ends the call on receiving a BYE request during an unconfirmed dialog. Previously in this scenario, SIP Server did not acknowledge the BYE request. (ER# 262426581)
SIP Server now correctly applies the Network Asserted Identity mechanism to 3pcc TMakePredictiveCall requests. Previously, SIP Server did not include the P-Asserted-Identity header in outgoing INVITE messages when processing 3pcc TMakePredictiveCall requests. (ER# 263094169)
Note: See also ER# 263094169 in the Known Issues section.
A new Application-level configuration option has been introduced:
sip-respect-privacy
Default Value: false
Valid Values: true, false
Changes Take Effect: At the next call
This option specifies what content SIP Server will report in the AttributeANI extension when an inbound INVITE message contains P-Asserted-Identity and Privacy:id headers. If this option is set to false, the content for AttributeANI is taken from the From header. If this option is set to true, the content for AttributeANI is taken from the P-Asserted-Identity header. (ER# 263638221)
The out-of-service mechanism now works correctly on DNs. Previously, after enabling a DN, SIP Server distributed an EventDNBackInService but did not use the DN until after the application was restarted. (ER# 263220547)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.05. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
In a scenario where TLS is not configured in SIP Server and SIP Server receives a request containing the Contact header with the transport=tls and the Record-Route header without transport=tls, SIP Server will now use the transport from the Record-Route header. Previously in this scenario, SIP Server rejected REGISTER, INVITE, and SUBSCRIBE requests. (ER# 262614644)
SIP Server now provides a busy tone to a caller if the caller is located behind�a softswitch. Previously, SIP Server did not apply any busy treatments in this configuration. (ER# 261976902)
This release incorporates TSCP release version 8.0.101.05 that corrects the following issue:
When SIP Server receives a TUnregisterAddress request for a DN of type VSP (TAddressTypeVSP), SIP Server now correctly sends an EventUnregistered message to its client. Previously, SIP Server mistakenly sent EventUnregistered in response to any request associated with this type of DN. (ER# 260676521)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.04. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
A new Application-level and DN-level configuration option, sip-from-pass-through, has been added to correct a compatibility issue with release 7.5. Starting with release 7.6, SIP Server replaces the username of the From header in the outgoing INVITE message if the use-contact-as-dn option is set to true.
sip-from-pass-through
Default Value: false
Valid Values: true, false
Changes Take Effect: For the next call
This option specifies whether SIP Server will use the content of the From header from the original INVITE to generate the content for the From header in the outgoing INVITE message. (ER# 261003309)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.04. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly resets the expiration timer after terminating a registration in cases where the REGISTER request contains the header Expires: 0 or the parameter Expires=0 in the Contact header (even if the Expires was included in a different dialog). Previously, after terminating the registration in this scenario, SIP Server sometimes did not reset the timer, causing DNs to be incorrectly placed in the out-of-service state. (ER# 261673075)
SIP Server no longer incorrectly uses the INVITE with Replaces method in certain routing scenarios. Now, the re-INVITE method is used instead. (ER# 261923728)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.04. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly passes all instances of UserData from T-Library to SIP messages, in cases where the UserData was attached separately at different points in the call. Previously, only the first attached UserData was mapped from the T-Library to SIP messages. (ER# 260851216)
SIP Server now correctly prints the command line parameter -nco to the SIP Server log. Previously, SIP Server sometimes incorrectly hid this parameter when printing to the log. (ER# 254140975)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.04. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly applies the option resource-allocation-mode when set to the value of circular, as described in the documentation. Previously, external routing resources were not allocated in a circular manner. (ER# 259592294)
SIP Server can now perform the alternate call operation to a call at a Routing Point from a SIP endpoint for which the sip-cti-control option must be set to hold. (ER# 214700427)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.03. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
When a predictive call is made to a busy destination (a dialog with the Media Server is initiated and an INVITE is sent to the customer) and the Media Server reports a cpd.busy result, SIP Server now correctly reports CallStateBusy (6). Previously in this scenario, SIP Server reported CallStateAllTrunksBusy (10). (ER# 259299936)
SIP Server now correctly processes the following scenario:
routing-timeout expires, the call is diverted to the ACD Queue (default-dn).INVITE to the caller.INVITE transaction times out in SIP Server and it drops the call.
If SIP Server receives a 1pcc re-INVITE with the Replaces header from one of the two parties on the call when both agent and caller greetings are in progress, SIP Server no longer drops the call. (ER# 250790141)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.03. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
The registrar-default-timeout option now works as documented in the 8.0.400.28 Release Note. Previously, SIP Server incorrectly calculated the expiration timeout for a REGISTER request if the Expires header was absent in the REGISTER request and the registrar-default-timeout was set to a value other than 0 (zero). (ER# 260578814)
When SIP Server routes a call with creation of the �Dummy� SDP in the INVITE request and the destination responds with an error, SIP Server now reports an EventError message set to the same error code as it would have been reported when routing a regular call. For example, if in this scenario the destination responds with a 486 Busy Here message, SIP Server would distribute the EventError set to the 231 code (DN is Busy). Previously, SIP Server distributed the EventError set to Unknown. (ER# 259938901)
While routing a call with creation of the �Dummy� SDP, SIP Server now uses only even numbers for RTP ports. The port number could be equal to a value of sdp-m-low but will be always less than sdp-m-high in AttributeExtensions of a TRouteCall request. Previously, when SIP Server used even and odd numbers as RPT ports, the messages with the odd numbers were rejected by the IP PBX. (ER# 260565932)
SIP Server now distributes the information sufficient to correctly display a list of conference parties on the desktop when a remote party is added to the conference. Previously, SIP Server incorrectly displayed conf=<Conference ID> on the desktop of the remote conference party. (ER# 259560650)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.03. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature or functionality.
The registrar-default-timeout configuration option has been modified as follows:
registrar-default-timeout
Default Value: Empty string
Valid Values: 0–4294967295
Changes Take Effect: At the next REGISTER dialog
This option specifies the expiration timeout for a REGISTER request as a value (in seconds)
in the 200 OK response that is sent by SIP Server to the SIP endpoint. When the option is set to 0, or is not defined, the Expires header value from the REGISTER request is used as the expiration timeout. If the option is set to any value other than 0, the timeout is set to the lesser of the option value and the value specified by the client.
(ER# 260029948)
This release includes the following corrections and modifications:
SIP Server again correctly dynamically updates DN-level option changes if the connection to Configuration Server is lost and then restored. In versions 8.0.200.34 to 8.0.400.25, this functionality was absent, and SIP Server had to be restarted for the DN-level option changes to take effect. (ER# 254606863)
SIP Server again correctly processes the pullback transaction type. In version 8.0.400.25, the pullback transaction type was not processed properly. (ER# 259826552)
Supported Operating Systems
New in This Release
Corrections and Modifications
There are no restrictions for this release. This release contains the following new features or functionality.
SIP Server now supports SIP traffic monitoring for enhanced reliability. SIP Server monitors incoming SIP traffic, and can initiate a switchover after a configurable length of time where no SIP messages are received. A new Application-level configuration option, sip-pass-check, must be configured to enable this functionality. In addition, at least one service device should be configured for Active Out-Of-Service Detection with oos-check and oos-force options.
sip-pass-check
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
This option enables tracking of SIP messages that reach the primary SIP Server, including responses from SIP devices (DNs) monitored by SIP Server using the oos-check and oos-force options.
The primary SIP Server summarizes results of the checks on DNs for out-of-service status and monitors the time passed since the last received SIP message. If the primary SIP Server does not receive SIP messages for a certain period of time, SIP Server reports the SERVICE_UNAVAILABLE status to LCA/SCS. The period of time is chosen as the maximum of sums (among the sums of the oos-check and oos-force option values, configured for service DNs). When SIP Server reports the SERVICE_UNAVAILABLE status to LCA/SCS, SCS switches the primary SIP Server over to the backup server, and it reports the SERVICE_RUNNING status to LCA/SCS. The backup SIP Server becomes the primary, and starts monitoring SIP traffic.
If both the primary and backup servers receive no SIP traffic, a switchover would occur each time the effective out-of-service timeout expires. To prevent frequent switchover in this case, SIP Server detects the "double switchover" condition and doubles the effective out-of-service timeout each time the double switchover happens, up to two times, or until one of the two servers detects SIP traffic. As soon as SIP traffic is detected, the server that detected the traffic remains the primary SIP Server and continues normal operation.
(ER# 252146303)
SIP Server now supports Video Contact Center with Media Server as follows: Video on Hold/Queue, Push Video, and Video Conference with active speaker detection.
SIP message mapping (from SIP to T-Library and from T-Library to SIP) is enhanced for Unicode support. SIP Server can convert UTF-8 encoded data received in the SIP messages to a local character set. Reverse conversion (from a local character set to UTF-8) is performed by sending call attached data or AttributeExtensions of a TRouteCall request encoded in a local character to the remote destination using SIP messages. A new Application-level configuration option, encoding, must be configured to enable this functionality.
encoding
Default Value: Empty string
Valid Values: See the ICU Home > Converter Explorer page for values (http://demo.icu-project.org/icu-bin/convexp)
Changes Take Effect: After SIP Server restart
This option activates Unicode support for the SIP-to-TLib and TLib-to-SIP mapping functionality (see the �Mapping SIP Headers and SDP Messages� section in the Framework 8.0 SIP Server Deployment Guide). By default, Unicode support is disabled. To activate this functionality, set this option to the name of a converter that can translate UTF-8 data to the local character set. The converter suitable for a particular deployment can be found using the ICU Converter Explorer.
(ER# 252146229)
SIP Server now supports sending �Dummy SDP� in the initial SIP INVITE message sent in certain call routing scenarios. To support this feature, SIP Server introduces the following key-value pairs in AttributeExtensions of a TRouteCall request:
| Key | Value |
|---|---|
| sdp-c-host | Any string. The value will be propagated as the connection address in the c= line of Dummy SDP. |
| sdp-m-port-low | Any integer that represents a valid UDP port. |
| sdp-m-port-high | Any integer that represents a valid UDP port. |
| after-routing-timeout | An integer that overrides the value of the after-routing-timeout configuration option, which specifies the length of time (in seconds) that SIP Server waits before diverting the call from the Routing Point DN to the destination DN after RequestRouteCall was processed. When the call is not diverted before the timeout expires, SIP Server generates the EventError message. |
sdp-m-port-low and sdp-m-port-high pairs are range boundaries. For each outgoing INVITE message, SIP Server will choose a different port from this range and include it as an RTP port in the m= line. If the sdp-m-port-high key is not specified (empty), the value of the sdp-m-port-low key will be used.sip-treatments-continuous=true
sip-enable-100rel=true
divert-on-ringing=falseSIP Server now supports dynamic DN replacement within SIP headers, which improves integration with other SIP elements. To support this feature, SIP Server introduces the new Application-level configuration option, tlib-map-replace-dn.
true, falsefalse[dn] pattern in SIP headers mapped from T-Library attributes. If you set this option to true, SIP Server replaces the [dn] pattern in mapped SIP messages with the digits of the DN where the SIP message is being sent. This applies to both AttributeExtensions mapping in TRouteCall, and UserData mapping as configured on a particular DN.
Note: This [dn] pattern replacement functionality applies to SIP header mapping only, not to Request-URI parameters mapping.
(ER# 257249490)
SIP Server now supports staggering of Busy Lamp Field (BLF) SUBSCRIBE messages. To support this feature, SIP Server introduces the new Application-level configuration option, subscription-delay.
subscription-delay
Default Value: 0
Valid Values: 0–10000
Changes Take Effect: Immediately
This option specifies the time interval (in milliseconds) between the new individual SUBSCRIBE requests used to create new SUBSCRIBE dialogs that SIP Server sends if several Voice over IP Service objects are configured with service-type set to blf.
Note: Genesys recommends setting the option to a value in a range of 20–200.
(ER# 256933954)
SIP Server now supports the routing of an outbound call to an external destination by using the REFER method. When the feature is activated, SIP Server places itself in the Out Of Signaling Path. To support this feature, configure the following options for a DN of type Trunk:
oosp-transfer-enabled=true
refer-enabled=true
(ER# 215121463)
SIP Server now supports RHEL 5.4.
SIP Server now supports VMware ESXi 4.1.
This release includes the following corrections and modifications:
SIP Server no longer sets a DN to out of service in a scenario where a call is routed to an unresponsive device and a caller abandons the call before the sip-invite-timeout timer expires. If the caller does not abandon the call during the sip-invite-timeout time period, then, when this timeout expires, SIP Server sets the unresponsive device to out of service. Once the recovery-timeout timer configured for this device expires, SIP Server sets it back in service. (ER# 102209228)
SIP Server now sends an EventError message in response to a TRetrieveCall request that fails because of the 481 Call/Transaction Does Not Exist response received from the CTI SIP endpoint to a NOTIFY (Event: talk) request. Previously, SIP Server ignored such a response and did not perform any action. (ER# 115340466)
In a scenario where a routing destination responds with a 486 Busy Here message and a call remains on the Routing Point, SIP Server no longer generates a duplicated EventError message that does not contain a ReferenceID attribute. (ER# 117313496)
SIP Server no longer creates a new session with Stream Manager when a PlayAnnouncementAndDigits treatment is requested with the following parameters: MAX_DIGITS is set to 1 and PROMPT is set to INTERRUPTABLE. (ER# 121568983)
SIP Server now begins the START_TIMEOUT timer when the announcement is finished and the collect digits session is started in the PlayAnnouncementAndDigits treatment. Previously, SIP Server began the START_TIMEOUT timer after receiving digits during the prompt play. (ER# 123089090)
SIP Server now sends an EventMonitoringCancelled message for a failed TMonitorNextCall transaction in a scenario where an ACD supervisor submits the TMonitorNextCall request to SIP Server and the supervisor�s endpoint does not respond to the SIP Server's INVITE requests. Previously, SIP Server re-transmitted INVITE requests, and then aborted the transaction by sending an EventError message without having the ReferenceID attribute in it. (ER# 155205348)
SIP Server now supports RFC 2617 and does not send the nonce-count (nc) parameter in the Authorization header if the server did not send a qop directive in the WWW-Authenticate header in the 401 Unauthorized response. (ER# 162834521)
SIP Server now correctly takes UserData from TSingleStepConference requests and includes it in EventAttachedDataChanged messages. (ER# 187414161)
SIP Server now rejects a TSingleStepTransfer request to a DN that is already involved in the call with an Invalid Called DN error message (error code 71). Previously in this scenario, SIP Server incorrectly sent a DN is not Configured in CME error message (error code 59). (ER# 193002692)
SIP Server now correctly generates a 223 Bad parameter passed to function error message when a TSendDTMF request is rejected because of an invalid input. Previously, SIP Server incorrectly generated an 1143 Object not known error message. (ER# 199744931)
When an agent initiates a 1pcc transfer (using the REFER method) to an unresponsive device, but the caller drops the call before the transfer is initiated, SIP Server now correctly terminates dialogs with related devices and releases all of the call-related resources. Previously, SIP Server did not release resources related to the dialog with the device of the transferring agent, which sometimes resulted in increased memory utilization. (ER# 203057205)
SIP Server now correctly processes 481 Call/Transaction Does Not Exist response messages. Previously in some rare cases, the 481 response sometimes resulted in a stuck call in SIP Server that might have led to excessive memory utilization and SIP Server instability. (ER# 216113388)
In a scenario where, after URS routes a call to Agent 1, Agent 1 tries to redirect the call to a busy Agent 2 (486 Busy Here message is received), SIP Server now sends an EventError message in response to the TRedirectCall request to notify that redirection has failed. Agent 1 now may decide whether to redirect the call to another agent or answer the call. Previously in this scenario, SIP Server incorrectly released the call. (ER# 219713401)
SIP Server now correctly sends a 180 Ringing message to a caller device during a 1pcc call if the 183 Session Progress message is received first from the destination device. (ER# 219714913)
SIP Server now processes OPTIONS messages as follows:
200 OK to an OPTIONS message within an existing active dialog, in order to support the keep-alive mechanism.OPTIONS message with a 481 Call/Transaction Does Not Exist when it receives the OPTIONS message for an established dialog (with the To, From, or Call-ID tags) which does not exist.200 OK to an OPTIONS message that is not part of an established dialog (no To tag)SIP Server now correctly processes scenarios where, during a call transfer, the destination device responds with a 408 Request Timeout error and the transferred party on the main call terminates the call. Previously in this scenario, SIP Server sometimes became unstable. (ER# 224290416)
SIP Server now correctly sets After Call Work (ACW) time for Agent 2 in the following scenario:
use-data-from is set to consult-user-dataconsult-user-data is set to inheritedmerged-user-data is set to merged-onlyUserData is updated with WrapUpTime=30 for the main call, then the call is routed to Agent 1.UserData is updated with WrapUpTime=60 for the consultation call, then the call is routed to Agent 2.NotReady (AfterCallWork).Ready after only 30 seconds instead of the required 60. (ER# 224983335)
If a previous Authentication-based challenge is rejected with a 401 Unauthorized message that contains a new challenge (for example, at renewal authorization time), SIP Server now correctly makes a new attempt: SIP Server re-sends the REGISTER request one more time, according to the new challenge received in the 401 message. (ER# 225901325)
SIP Server now correctly includes the host in the URI for the From header in SUBSCRIBE requests. In addition, SIP Server also sets the default Expires value to 600. Previously, the host was sometime missing and the Expires value was set to the incorrect default of 3600. (ER# 229052321)
SIP Server now correctly generates EventRouteUsed messages in the following scenario:
TRouteCall request with an empty destination field, a call is redirected to an ACD Queue configured as the default-dn.180 Ringing response, SIP Server now correctly generates the EventRouteUsed according to the setting divert-on-ringing=true. Previously in this scenario, SIP Server did not send the EventRouteUsed to its T-Library subscribers. (ER# 229108894)
SIP Server now retries treatments only on media servers that are still in service (the out-of-service check shows the Voice over IP Service DN (service-type set to treatment) as available). (ER# 230151967)
If a device fails to respond to the BYE messages that SIP Server sends during a single-step transfer (using the REFER method) of one external party to another external party, SIP Server now sends BYE messages up to a timeout of 32 seconds, after which it releases all resources related to the dialog with the device. This applies in cases where the Trunk DN representing the transferred device is configured with the options refer-enabled=true and oos-transfer-enabled=true. Previously, SIP Server did not release resources related to the dialog with the unresponsive device, which sometimes resulted in increased memory utilization. (ER# 235421980)
SIP Server now properly executes mapping of SIP header parameters into T-Library event messages. Previously, if the SIP parameter value was presented as a quoted string containing spaces, SIP Server would only include the content up to the first space, leaving out all the content after that. (ER# 235859001)
When processing the RecordUserAnnouncement treatment, SIP Server now adds only one USER_ANN_ID parameter in AttributeExtensions when generating an EventTreatmentEnd message. (ER# 238866082)
SIP Server now correctly processes calls initiated by an incoming INVITE where the user part of the URI in the Contact header matches the number of any configured DN. Previously, if the user part of the Contact URI matched the number of a configured Routing Point or ACD Queue DN, SIP Server rejected the INVITE with the 603 Decline response even if the domain part of the URI was listed in the enforce-external-domain option. (ER# 239012316)
SIP Server now correctly processes the following scenario:
UserData is attached to the call. The userdata-map-filter option is set to pass UserData in the SIP message.INVITE message to GVP containing UserData attribute values in the X-Genesys custom headers. BYE message.BYE message as UserData. If same key is repeated in the header as well as in the BYE message, then preference will be given to the BYE message and the original key-value pair will be replaced in the UserData.
SIP Server now correctly sends an EventTreatmentNotApplied message for a failed TApplyTreatment request in the following scenario:
INVITE message with a 486 Busy Here response. SIP Server now correctly finalizes unsuccessful outbound transfers via the trunk (configured with the reuse-sdp-on-reinvite option set to true) if the transfer controller device is configured as a single-dialog device (the dual-dialog-enabled option is set to false). SIP Server now releases the consultation call on the agent device after the ringing timeout expires and properly deletes the consultation call. Previously, SIP Server did not handle unsuccessful outbound transfers via the trunk correctly and released both main and consultation calls.(ER# 244213466)
In high-availability environments, No-Answer Supervision now works for calls ringing during a switchover. (ER# 244480369)
When operating with Cisco UCM, Genesys recommends configuring the following options for the Voice over IP Service DN (service-type set to softswitch) that points to the UCM node:
make-call-rfc3725-flow = 2
ring-tone-on-make-call = true
refer-enabled = false
sip-ring-tone-mode = 1
dual-dialog-enabled = false
The reuse-sdp-on-reinvite option must not be used. Previously, Genesys recommended setting the make-call-rfc3725-flow to 1, which sometimes caused an issue with the audio path (no audible ringback) for the SCCP devices. (ER# 245634993)
The Max-Forwards header is now populated in all request messages, including CANCEL messages. (ER# 246018341)
When an outbound call is initiated from a device, SIP Server no longer drops the nailed-up connection after receiving a 486 Busy Here message from the destination device. (ER# 246231860)
A new Application-level configuration option, send-200-on-clear-call, has been added.
send-200-on-clear-call
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
When this option is set to true, SIP Server, when executing a TReleaseCall request for a non-established call, terminates the call leg in the dialing state by sending a 200 OK message. When this option is set to false, SIP Server sends a 404 Not Found in this scenario.
(ER# 246282016)
SIP Server now correctly processes the following scenario:
REFER method.NOTIFY message containing Subscription-State: terminated to the transferring device.NOTIFY message, SIP Server receives a BYE message from the transferring device.BYE messages properly and releases dialogs related to the transferring device only. Previously, SIP Server erroneously terminated the SIP dialog with the transfer destination, which at that time was connected to the caller. (ER# 247072953)
SIP Server now correctly processes transfer scenarios where the transferring party invites the destination party, but the destination responds with a provisional response followed by an error. Previously, when the transfer destination party sent an error message, SIP Server did not terminate the dialog, resulting in a stuck call. (ER# 247204273)
SIP Server now rejects any NOTIFY message that is not part of the existing dialog by sending a 481 Call/Transaction Does Not Exist message in response, except when the NOTIFY is sent regarding the Message Waiting Indicator (MWI) feature. (ER# 229883676, 247824702)
SIP Server now supports the full range of boolean values for the option nas-private at the DN/Agent Login level: true/false, 1/0, yes/no, and on/off. All other values are ignored. Previously, only the values 0 and 1 were accepted. (ER# 247952524)
In multi-site environments, when SIP Server 1 transfers a call using the REFER method, SIP Server 2 now adds a Referred-By header in the new INVITE message. This allows SIP Server 1 to match the new INVITE with the transferred call and properly release it. Previously, SIP Server did not add a Referred-By header, which sometimes resulted in a stuck call in SIP Server 1, which originated the transfer. (ER# 248405051)
SIP Server now correctly releases a call when it receives a 503 Service Unavailable message in response to a re-INVITE request that it sent to the call originator. (ER# 248405320)
SIP Server now correctly cleans up unsuccessful outbound calls initiated by an agent with a nailed-up connection. Previously, after the unsuccessful outbound call timed out, the agent would become inaccessible. Now, the agent remains in the correct state after an unsuccessful outbound call. (ER# 248848499)
HA SIP Servers now synchronize all dialogs, including those with unacknowledged transactions. Previously, HA SIP Servers did not synchronize these dialogs properly, which sometimes resulted in errors in subsequent transactions in the backup SIP Server after a switchover occurred. (ER# 248991280)
SIP Server now does not announce the early media support when initiating a connection with Genesys Media Server (does not send the Supported: 100 rel header in the INVITE to the Media Server if it is configured as a DN of type Voice over IP Service in the configuration environment). Previously, this early media support sometimes caused SIP Server to incorrectly fail the call. (ER# 249538361)
SIP Server now supports No-Answer Supervision (NAS) for DNs in the following consultation call scenarios:
nas-private is set to true. (ER# 250053339)inherit-bsns-type is set to true. (ER# 250144793)SIP Server now correctly processes Hold Call in multi-site scenarios. Previously in some multi-site scenarios, SIP Server sometimes incorrectly send a new INVITE request through an existing dialog where a previous INVITE transaction was not yet finished. (ER# 251273998)
SIP Server no longer gives up trying to open the SIP listening port after a failed attempt. To solve this issue, SIP Server introduces the new Application-level option sip-max-retry-listen.
150-65535(ER# 253914415)
SIP Server now correctly adds the GVP-Resource-ID header in the request URI for both INVITE and NOTIFY requests that it sends to GVP. (ER# 253926886)
In cases where an Agent Login object is configured with an empty password, SIP Server now correctly generates an EventAgentLogin message without authentication exchange for Polycom IP phones that support agent-status updates initiated from the device. Previously, when SIP Server received SUBSCRIBE messages from these devices, it incorrectly responded with a 406 Not Acceptable message. (ER# 254190923)
When a treatment is applied to a call at a Routing Point and the call is routed to an external destination over a trusted trunk (enforce-trusted=true), SIP Server now correctly adds the P-Asserted-Identity and Privacy headers into outgoing INVITE messages. (ER# 254582229)
When an agent becomes unavailable with a specified reason code, SIP Server no longer reports the agent reason code in subsequent agent-related events. SIP Server now reports the correct agent-related events, without adding a reason code. (ER# 254530610)
SIP Server now correctly processes scenarios where a party is deleted during the processing of a T-Library request. Previously in this situation, SIP Server sometimes became unstable. (ER# 256975374)
SIP Server now correctly processes SIP 302 Moved Temporarily responses in cases where the domain part of the SIP URI passed in the 302 Contact header was not recognized as an internal domain. Previously in this scenario, SIP Server would create an incorrect Request URI for the outgoing INVITE by adding a suffix .<DOMAIN-NAME> to the user part. As a result, the remote destination could not identify the target and rejected the INVITE. (ER# 253867444)
SIP Server now successfully completes transfers in multi-site environments in cases where the main call is being recorded. Previously, SIP Server sometimes released all parties during an attempt to complete such a transfer. (ER# 231902069)
SIP Server now successfully reports the transferred DN as the AttributeANI in the EventPartyChanged message when completing transfers in multi-site environments. Previously, SIP Server sometimes inadvertently set the transferring party DN as the AttributeANI. This issue occurred when the option inter-server-trunk was set to true on the trunk of the destination SIP Server, pointing to the originating SIP Server. (ER# 224048754)
SIP Server now correctly rejects INVITE requests when the Contact header contains the wrong URI (host name followed by a colon, but the port is missing). Previously, SIP Server accepted these INVITE requests, but was later unable to send any requests within the dialog, resulting in stuck calls. (ER# 237670675)
SIP Server now supports messages in which a single Via header contains a comma-separated list of addresses. (ER# 245558626)
SIP Server now correctly handles the following scenarios:
TRedirectCall request is received when a call is ringing while still at a Routing Point (in this case, the option divert-on-ringing is set to false).
SIP Server now distributes the reason private-call inside the AttributeExtensions in cases where the Application-level option reason-in-extension is set to true. Previously, SIP Server always distributed this reason inside the AttributeReason. (ER# 254860258)
For predictive calls to be routed correctly in IMS deployments, you no longer required to configure the SIP Server application with the option p-asserted-identity, using the same value for the asserted identity as used for the Routing Point DN.
(ER# 254989473)
SIP Server now generates an EventRetrieved message in response to a TAlternateCall request in scenarios where GVP is configured as an MSML Media Server. (ER# 250926651)
SIP Server now supports REFER requests sent during the Early Dialog state. (ER# 249508823)
SIP Server now works properly if its connection with an MCU fails in scenarios involving call recording. (ER# 249358013)
If an attempt to add a new participant to a conference fails, SIP Server now checks if an alternate MCU is available. If available, then SIP Server will release the existing MCU and it will try to establish the conference using the alternate MCU. (ER# 250802005)
SIP Server now correctly applies Calling Line Identity Restriction (CLIR) on each dial-plan request to support the privacy requirements described in RFC 3325. In this case, Anonymous is used to the From, P-Asserted-Identity, and Privacy headers, as applied by the CLIR requested by the dial-plan. If clir is set to on, SIP Server updates the From in TMakeCall requests only. (ER# 235360404, 248341724)
In certain scenarios, if an HA switchover takes place while an
INVITE transaction was in progress, SIP Server now sends the
CANCEL request to the same location as where the original
INVITE was sent. Previously, this issue occurred in the following
scenarios:
request-uri option for the DN is configured with a
different host than the host configured in the contact option
on the same DN.INVITE request
contains a Contact header with a URI that is different from the
INVITE destination.(ER�233704551)
SIP Server is now able to dynamically update the DN-level options
sip-enable-moh and request-uri. (ER# 221245281)
SIP Server can now automatically release a monitored call when the following conditions are true:
MonitorScope is set to call and
MonitorMode is set to coach.To complete this call, either the second agent or the supervisor should release the call. (ER# 185899321)
In order to correctly process 1pcc calls to a PSTN destination in Alcatel IMS deployments, the following options no longer necessary to configure on the originating DN:
make-call-rfc3725 to a value of 2.sip-ring-tone-mode to a value of 1.When sending a REFER message to the other party, SIP Server no longer incorrectly sends a Max-Forwards value of 6, instead of the expected value as described in the RFC. (ER# 229815568)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.06. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
The setting of the X-Genesys-Orig header has been modified in the following scenario:
TReleaseCall request.BYE request sent to Agent A, the X-Genesys-Orig header is set to Agent B.Previously, in the BYE request sent to Agent A, the X-Genesys-Orig header was set to Agent A. See also the Release Note for version 8.0.300.62. (ER# 271148000)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.06. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now handles the setting of the X-Genesys-Orig header differently in the following scenario:
TReleaseCall request.
BYE request sent to Agent B, the X-Genesys-Orig header is set to Agent A. Previously, the header would be set to Agent B instead.
Note: If Agent A performs the TReleaseCall, the X-Genesys-Orig header will be set to Agent A, as it was previously. The change in value only applies when Agent B releases the call.
(ER# 271148000)
The out-of-service mechanism now works correctly on DNs. Previously, after enabling a DN, SIP Server distributed an EventDNBackInService but did not use the DN until after the application was restarted. Also, SIP Server can again dynamically update DN-level option changes if the connection to Configuration Server is lost and then restored. In versions 8.0.200.34 to 8.0.400.25, this functionality was absent, and SIP Server had to be restarted for the DN-level option changes to take effect. (ER# 272600387)
SIP Server now correctly processes the second attempt to route a call in multi-threaded configurations. Previously, SIP Server sometimes failed the second route attempt with the EventError DN is busy. (ER# 271367403)
SIP Server no longer terminates unexpectedly when the after-routing-timeout option expires in multi-threaded configurations. (ER# 271936350)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.06. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now includes the correct spelling for the header Content-Type: application/sdp in all cases. Previously in some scenarios, a misspelling in this header could cause SIP Server to inadvertently drop the call. (ER# 270426125)
SIP Server now correctly processes user data in the following scenario:
UserData.
UserData in the X-Genesys custom headers of the INVITE that it sends to GVP (SIP Server option userdata-map-filter is set to UserData).
BYE message that it sends to SIP Server.
BYE message as UserData. If the same key is repeated in both the header as well as in the BYE message, then preference is given to the BYE message; the original key-value pair will be replaced in the UserData.
(ER# 269814806)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.101.06. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly handles scenarios where an outbound call is released by a caller using a RequestReleaseCall, while SIP Server receives only a 100 Trying message in response to the INVITE; after that, the caller makes a second call to the same destination and SIP Server receives a 486 Busy Here message in response to the INVITE. Previously in this scenario, SIP Server could become unstable. (ER# 266904121)
SIP Server no longer becomes unstable when the transfer-destination party in a routing scenario is found to be Out-of-Service. (ER# 266981128)
SIP Server now successfully completes TCompleteTransfer operations after it receives a 500 Internal Server Error in response to a re-INVITE with media server SDP for music-on-hold. Previously in this scenario, SIP Server could abort the forthcoming TCompleteTranfer with an error. (ER# 267159412)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now correctly handles scenarios where the call destination sends multiple reliable provisional responses. Previously in this scenario, SIP Server responded with PRACK only to the first provisional response received from the call destination, but did not send PRACK to acknowledge the subsequent responses. As a result, the terminating party dropped incorrectly from the call. (ER# 261827472)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release does not contain any new features or functionality.
This release includes the following corrections and modifications:
SIP Server now successfully processes INFO messages sent during INVITE transactions where early media is established. Previously in this scenario, the 200 OK in response to the INVITE was ignored. (ER# 260758654)
180 Ringing responses that do not have any SDP are now skipped as soon as early media is established. Previously, 180 Ringing responses with 100rel and empty content were delivered to the other dialog with inserted SDP content and 100rel, which could confuse the destination device. (ER# 261397320)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature.
In transfer scenarios, where SIP Server is not engaged in signaling related to further REFER handling, SIP Server uses the content of the NOTIFY request that it receives from the destination for transfer completion. Prior to this release, SIP Server completed REFER-based transfers when it received a 200 OK message in the body of the NOTIFY request. A new DN-level configuration option, sip-transfer-complete-message, enables control of SIP Server behavior for completion of REFER-based transfers.
sip-transfer-complete-message
Default Value: An empty string
Valid Values: 1XX, 18X, 180, 183
Changes Take Effect: Immediately
Defined for a DN through which SIP Server sends a REFER request, this option specifies on which message SIP Server completes REFER-based transfers.
When this option is set to:
REFER-based transfer after receiving a 200 OK message within the NOTIFY request (the default behavior).1XX—SIP Server completes the REFER-based transfer after receiving any provisional response within the NOTIFY request.18X—SIP Server completes the REFER-based transfer after receiving a 180 or 183 provisional response within the NOTIFY request.180—SIP Server completes the REFER-based transfer after receiving a 180 Ringing provisional response within the NOTIFY request.183—SIP Server completes the REFER-based transfer after receiving a 183 Session Progress provisional response within the NOTIFY request.(ER# 257255267)
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 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new feature.
SIP Server now supports DN-level configuration for the options sip-enable-sdp-codec-filter and audio-codecs. If sip-enable-sdp-codec-filter is set to true in the DN configuration, SIP Server, as it propagates the SDP to and from the device represented by this DN, will use as its list of available codecs the value configured in the audio-codecs option on the DN rather than on the application. If sip-enable-sdp-codec-filter is set to true at both the application and the DN level, then the audio-codecs configured in the DN should contain a subset of the audio-codecs configured in the application. (ER# 256534025)
This release includes the following corrections and modifications:
SIP Server now correctly processes scenarios where a call recovery attempt fails and the call is deleted during the processing of a T-Library request. Previously, SIP Server sometimes became unstable in this scenario. (ER# 253363042)
SIP Server now synchronizes agent states in High Availability (HA) deployments, in cases where the backup SIP Server is started after a change to an agent state. Previously, when the backup SIP Server connected to the primary SIP Server, agents states were not properly synchronized.(ER# 256205854)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 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 generates EventResourceInfo messages (used to report on the availability of ports allocated for the outbound campaign) in cases where the subscription with GVP has expired. Previously, if SIP Server received the SIP NOTIFY from GVP before the 200 OK response to the original SUBSCRIBE request, SIP Server was unable to renew the subscription with GVP, and the EventResourceInfo message was not generated. Now, SIP Server consistently generates EventResourceInfo by refreshing GVP subscription in timely manner.
(ER# 255082493)
SIP Server now properly routes REFER-based transfers in cases when a 100 Trying is sent in response to the REFER message. Previously, SIP Server generated an error on receiving the 100 Trying in this scenario. The problem occurred when the destination Trunk DN was configured with oosp-transfer-enabled set to true. (ER# 256039171)
SIP Server no longer exits unexpectedly when an agent submits a TRetrieveCall request for a primary call while the consultation call is active. Certain SIP phones (for example, Microsoft RTC), when configured with dual-dialog-enabled set to true, cannot handle this scenario properly. Previously in this case, if the agent sent a TInitiateConference or TSingleStepTransfer request when the EventRetrieved arrived, SIP Sever could unexpectedly exit when trying to process these requests. Now, SIP Server no longer exits in this scenario, allowing agents to release the call. (ER# 255625401)
SIP Server now correctly attaches the repeat= parameter when restarting a service on a second media server (in cases where the original media server failed). Previously, SIP Server sent the INVITE to the second media server without this parameter. (ER# 255766481)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 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 successfully sends an ACK message in cases where the external party takes longer than 32 seconds to answer a TMakePredictiveCall request. Previously in these scenarios, SIP Server was sometimes unable to send the ACK, resulting in a failure to properly connect the call. (ER# 255236008)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
Starting from this version, SIP Server supports Alcatel IP Multimedia Subsystem (IMS) 9.0.
This release contains the following corrections and modifications.
SIP Server now supports the ability to control reliable provisional responses from within a routing strategy. The strategy can now add the new key-value pair sip-enable-100rel to the Extensions attribute of the RequestRouteCall operation. If the value of this key in the call request is false, SIP Server does not place the 100Rel in the header for the corresponding INVITE. This prevents the destination DN from sending reliable provisional responses, so that the SDP in the provisional response does not force SIP Server to interrupt an ongoing voice treatment. The Extensions setting takes priority over the sip-enable-100rel option configured at the SIP Server Application-level. However, the Extensions setting does not take effect for calls distributed over Trunk DNs where the option sip-server-inter-trunk set to true. (ER# 254140585)
SIP Server now correctly adds custom headers, as configured in the userdata-map-filter option, to INVITE messages sent to the media server in cases where a previous treatment was successfully applied to a call. (ER# 254522048)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
SIP Server now supports an increased maximum ringing period (longer than 32 seconds) for predictive calls through Genesys Media Server. To enable a ringing period of greater than 32 seconds, you must configure the following:
Trunk Group or Voice over IP Service), set predictive-timerb-enabled to false. truetrue, falseACK is not sent to the Media Server within 32 seconds after the 200 OK is received.false, SIP Server disables this timer and instead times the call using the AttributeTimeout value included in the TMakePredictiveCall request. If this timeout expires before the call is answered, or if SIP Server receives a BYE message from the Media Server, SIP Server terminates the call. Genesys recommends setting the AttributeTimeout to a value greater than zero (0) to prevent inadvertent call termination.true, SIP Server uses the 32-second timer.sip.timer_si option to a value greater than the AttributeTimeout used by SIP Server to control the call. Typically this setting comes into effect for stuck calls only, or in cases where AttributeTimeout is set to 0. For more information about this new MCP option, see release 8.1.101.43 in the Voice Platform 8.1 Media Control Platform Release Note.sessmgr.acceptcalltimeout to a value greater than the sip.timer_si. This prevents the MCP application from interfering with the SIP level timers.This release contains the following correction.
SIP Server now consistently sends INVITE requests to the correct gateway (address specified in the contact option in the corresponding Trunk DN). Previously, SIP Server sometimes inadvertently sent the INVITE to the IP address for an incorrect gateway. This issue occurred when:
Trunk DN objects are configured with the option oos-check set to a non-zero value.contact option was modified).
Previously, SIP Server could consume 100% CPU usage when trying to convert a domain name of DN contact into an IP address in networks where no DNS server is available. This problem occurred when the switch configuration included Trunk or Voice over IP Service DNs with the oos-check option set to a non-zero value. This version of SIP Server corrects this problem. Also, a new application-level option to select a DNS resolution mode has been introduced.
sip-enable-gdns
Default Value: true
Valid Values: true, false
Changes Take Effect: After SIP Server restart
Specifies the DNS resolution mode. If you set this option to true, SIP Server uses its internal DNS client to connect to the DNS server available on the network, in order to use its conversion services. If no DNS server is available, set this option to false. In this case, SIP Server resolves the domain names using local operating system utilities.
If set to false, SIP Server is unable to perform DNS resolution for SRV records with contacts that are missing port information (indicating the need to use SRV). Instead, A record resolution and default ports will be used. The default port for UDP/TCP is 5060, while the default for TLS is 5061.
(ER# 253786977)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
The range of valid values for the subscribe-presence-expire configuration option has been modified. The option can now be set to a value of up to 259200 (seconds, which corresponds to 72 hours). Previously, the upper limit for this option was 3600 (1 hour). (ER# 253820313)
This release includes the following corrections and modifications:
When configured with option divert-on-ringing set to false, SIP Server now always postpones sending an EventRouteUsed message until the routing destination sends the final response. Previously, SIP Server distributed EventRouteUsed when the routing destination sent a provisional response with the message body containing SDP. (ER# 250320902)
SIP Server now correctly executes greetings in predictive call scenarios. Previously, SIP Server sometimes did not play greetings in certain predictive call scenarios, where recording and greeting services were scheduled for the call. (ER# 252498507)
SIP Server no longer drops the call if a CompleteConference attempt fails because Stream Manager terminates the dialog with a BYE request and the header Reason "No matching codecs found". (ER# 229946401)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This is a hot fix for this product. This release contains the following new features or functionality.
SIP Server now supports a new Application-level configuration option, subscription-delay.
subscription-delay
Default Value: 0
Valid Values: 0–10000
This option specifies the time interval (in milliseconds) between the new individual SUBSCRIBE requests used to create new SUBSCRIBE dialogs that SIP Server sends if several Voice over IP Service objects are configured with service-type set to blf.
Note: Genesys recommends setting the option to a value in a range of 20–200. (ER# 253900781)
This release includes the following corrections and modifications:
SIP Server now correctly processes MakeCall requests from an agent with a nailed-up connection to the contact center, in cases where the agent abandons the preceding call. Previously, if the agent abandoned a call in process, subsequent MakeCall requests sometimes resulted in an EventError. (ER# 252571986)
SIP Server now supports Hold/Retrieve operations on outbound calls originated from CTI phones (phones controlled by BroadSoft talk/hold extensions) when the SIP dialog to a destination is in an early state. In this case, the SIP dialog with the originating device must already be established, otherwise the THoldCall or TAlternateCall request is rejected with an EventError.
For CTI phones that support early media, set the refer-enabled option for its DN to false.
SIP Server now properly extracts part of the XML message related to a particular entity from the NOTIFY message that arrives from the BroadSoft�s BroadWorks switch. Previously, SIP Server sometimes extracted an incorrect part of the XML message, leading to inaccurate reporting of agent state updates. (ER# 253900786)
Supported Operating Systems
New in This Release
Corrections and Modifications
There are no restrictions for this release. This release contains no new features or functionality.
This release includes the following corrections and modifications:
SIP Server no longer undergoes performance degradation depending on the number of active calls and party activity. This release removes the storage of unnecessary objects, restoring SIP Server efficiency. The only consequence of this change is that when the option greeting-delay-events is set to all (resulting in postponed EventEstablished in greeting scenarios), SIP Server no longer distributes EventEstablished in cases where the media server responds with a 404 Not Found error to the first (customer-side) greeting INVITE. (ER# 251763723)
SIP Server now requests GVP to record the Call Progress Detection (CPD) phase of a predictive call if the CPD analysis is performed on the media gateway, when the TMakePredictiveCall request contains the cpd-record key-value pair set to on in the Extensions attribute. (ER 238357601)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.100.12. 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 SIP Server.
This release is under shipping control. This section describes new features that were introduced in this release of SIP Server.
Overload Control. SIP Server now supports an overload control mechanism, which can gracefully handle situations where the load on the SIP Server exceeds its configured rate capacity threshold(s). (ER# 242227518)
Alternate Routing for External Destinations. SIP Server can now route inbound 1pcc calls to a specified DN when the INVITE is addressed to an external destination. To support this feature, a new Application-level configuration option, default-route-point, has been added. (ER# 242227612)
Support for Genesys Media Server. SIP Server supports integration with the Genesys Media Server for providing media services including call treatments, call recording, call supervision and basic IVR.
Enhanced Support for Dial Plan. SIP Server supports additional functionality for the Dial Plan feature:
Network Asserted Identity. SIP Server supports the network asserted identity mechanism defined by RFC 3323, RFC 3324, and RFC 3325. (ER# 242227493)
Control of SIP Response Code from within a Routing Strategy. SIP Server supports user-specified error code responses for rejected calls, as applied in the Extensions attribute in call requests made from a routing strategy. (ER# 242227592)
Multi-Threaded Architecture. SIP Server supports multi-threading for enhanced scalability for machines with multiple CPUs. (ER# 242227520)
Application Failure Detection. SIP Server supports Management Layer monitoring of the SIP Server application, in order to take corrective action should the application or one of its threads fail. (ER# 242322523)
Remote Media on Genesys SIP Endpoint SDK 8.0. SIP Server supports remote 3pcc Beep Tones Control and DTMF Tones Control on custom endpoints built from the Genesys SIP Endpoint SDK 8.0. (ER# 242227532)
IMS Integration. SIP Server supports integration with IP Multimedia Subsystem (IMS) networks. (ER# 242227570)
Transport Layer Security (TLS) for SIP Traffic. SIP Server supports secure communication using TLS for the SIP listening port. (ER# 242227516)
Presence Integration for Microsoft OCS 2007 R2. SIP Server now supports the ability to subscribe to the presence status for Microsoft Office Communications Server 2007 R2 users. (ER# 242227578)
Enhanced Support for Personal Greeting. SIP Server now enables you to configure how personal greetings are provided in different scenarios (for example, after transfers or call conferences). (ER# 242322525)
SIP Server now supports the geo-location feature for consultation calls to external destinations. For consultation calls, SIP Server looks for an external destination based on the geo-location value as set on the originating DN for the consultation call. If
the consultation scenario involves a Routing Point, then SIP Server checks for geo-location according to the following priority (listed in order of precedence from highest to lowest):
Extensions parameter in the TRouteCall request.geo-location option as set on the Routing Point DN.geo-location option as set on the originating DN for the consultation call.Changes to order of service. SIP Server now processes call operations for an established call in the following order of priority:
(ER# 253954691)
This release includes the following corrections and modifications:
SIP Server now updates the sip-ip-tos configuration option setting dynamically. (ER# 238135561)
SIP Server can now disconnect clients that become unresponsive or stop processing T-Library events after a certain period of time. Previously, SIP Server did not disconnect such clients, which negatively affected SIP Server performance. (ER# 219632231)
In cases where the request RequestAgentLogin contains the attribute AttributeAgentWorkMode=1 (ManualIn), SIP Server will set the Initial state for the agent to NotReady, and in the reported EventAgentNotReady message, the AttributeAgentWorkMode will be correctly set to 0 (Unknown). Previously in this scenario, the AttributeAgentWorkMode was set to 1. This issue occurred if EventAgentNotReady was sent after RequestAgentLogin containing the AttributeAgentWorkMode set to 0 (Unknown) had arrived. (ER# 226919581)
SIP Server now properly completes a transfer triggered by a REFER from an agent phone by releasing the consultation call and keeping the main call. Previously, SIP Server mistakenly released the main call and kept the consultation call. This situation occurred when the consultation call was placed on hold before the transfer was completed.
(ER# 224639000)
Multi-site supervision is now correctly applied to a call if the cast-type configuration option is set to route. This enables SIP Server to correctly distribute calls from an external Routing Point, through ISCC, to a supervisor. (ER# 237105798)
SIP Server now consistently releases the external party from a call if that party initiates a transfer using the SIP REFER method. Previously, SIP Server kept the external party on the call, which could affect future call flow. In particular, SIP Server did not distribute EventCallPartyAdded if a REFER consultation call was made to the same external destination, followed by a RequestCompleteConference. (ER# 243280146)
SIP Server now allows using a full Windows path when configuring a recording file name. (ER# 129836621)
SIP Server now correctly processes the following scenario:
RequestMonitorNextCall to monitor the agent in the call.RequestCancelMonitoring while his/her device is ringing.EventMonitoringCancelled.CANCEL to the supervisor's device after the 32-second timeout expires.RequestSingleStepConference was received. EventMonitoringCancelled message but did not release the resources allocated for the unsuccessful monitoring session. (ER# 155205248)
SIP Server no longer treats an active call with a duration longer than one hour as stuck, unexpectedly releasing it. (ER# 242685937)
SIP Server now uses different timeout options for regular devices and media service devices, in order to correctly process scenarios where only a provisional response is received after sending an INVITE to a device (without receiving a final response). SIP Server treats the expiry of either timeout setting the same way it does an expiry of SIP Timer B.
sip-invite-timeout—For regular devices, used to specify the length of time that a SIP transaction can remain in the Proceeding state when the only provisional responses that it receives are 100 Trying messages. Any other provisional message removes the timer, so that the regular device can remain in a ringing state until the peer's action causes SIP Server to cancel the INVITE request.
.sip-invite-treatment-timeout—For media service devices, used to specify the length of time to wait for a final or reliable provisional response. If this timeout expires, the media service device is considered to be out of service and SIP Server tries to use an alternative device to perform the required function.Previously, this scenario might have resulted in stuck transactions, dialogs, and calls. (ER# 247497885)
SIP Server now supports the capacity and capacity-group options for DNs of type Voice over IP Service (with service-type set to softswitch). (ER# 240241639)
SIP Server now correctly terminates all dialogs initiated towards a recorder if the call is released before the dialogs with the recorder are established. Previously in this scenario, SIP Server did not send a BYE message to one of the recorder dialogs. (ER# 243365569)
SIP Server can now verify the DN name (for example, to check if a space is included). The new verify-sip-names configuration option has been added.
verify-sip-names
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Specifies whether SIP Server performs the DN name verification. If the option is set to true and the name is found to be incorrect, SIP Server will return the Incorrect address format error message. (ER# 202632713)
SIP Server no longer applies the recovery-timeout configuration option value if the sip-oos-enabled option is set to false in the same DN configuration. Previously, SIP Server processed both option settings, which resulted in misleading log printouts stating that the DN was set to out of service. (ER# 246057046)
At startup, SIP Server now goes to the Service Unavailable status and then to the Started status. This allows Solution Control Server (SCS) to clear a Service Unavailable alarm automatically at SIP Server successful startup. (ER# 232888891)
SIP Server now correctly distributes events when a GVP PlayApplication treatment is requested after a RingBack treatment. Currently, SIP Server distributes events as follows:
RingBack treatment is requested. SIP Server generates an EventApplied for the treatment.PlayApplication treatment is requested with GSIP_RM_URI in the treatment parameters.EventTreatmentEnd for the RingBack treatment and sends an INVITE for the PlayApplication treatment.EventApplied, EventAttachedDataChanged and EventTreatmentEnd messages after the 302 Moved Temporarily is received from GVP.SIP Server now correctly processes Predictive Call scenarios in which the call is dropped when the Paraxip gateway responds to the INVITE with a 200 Ok that contains CPD-Result: Answering-Machine. In this case, SIP Server first sends an ACK then a BYE to the gateway, in accordance with RFC 3261. Previously, SIP Server did not send the ACK. (ER# 240465645)
SIP Server now correctly merges transferred or conferenced calls when in Backup mode. (ER# 242030683)
SIP Server now correctly selects a gateway or trunk for the outbound call when the geo-location option is enabled. (ER# 234668569)
Use of the dial-plan parameter calltype no longer conflicts with the DN-level option override-call-type when configured on a DN of type Routing Point. Previously, combining the use of these two options might have resulted in CallType attribute changes midway through the call. (ER# 234758271)
SIP Server no longer inadvertently starts recording too early while processing TMakePredictiveCall requests. Previously, SIP Server might have started recording at the EventRouteRequest rather than at the moment when the call is established on the agent phone. This issue occurred when the record option was set to true on the Trunk DN that represents the outbound gateway. (ER# 229724438)
SIP Server no longer incorrectly updates capacity information when active calls are present on a trunk. Previously, Genesys recommended that you restart SIP Server after completing the capacity configuration. This is no longer necessary. (ER# 172370691)
SIP Server now correctly generates an EventEstablished message for DN1 in the following scenario:
A message with an empty body is no longer able to disrupt a chat session when using the Instant Messenger. (ER# 114869267)
As specified in RFC 3265, SIP Server now correctly deactivates a subscription in cases where a SIP device—other than Genesys SIP Endpoint—registers for a DN that has an active subscription created by Genesys SIP Endpoint. Previously, SIP Server incorrectly used the active subscription information and NOTIFY (talk) messages could be sent to a different IP address than the one used by the SIP device currently registered for the DN. (ER# 208999570)
SIP Server now correctly responds to INFO requests with a 481 Call/Transaction does not exist in cases where the internal registrar is disabled and the name of the external registrar is empty. Previously, SIP Server might have sent the wrong response, or no response. (ER# 229851405)
SIP Server now correctly sets AttributeCallType=3 (outbound) in the EventDialing in cases where a TMakeCallRequest is issued toward an external destination. Previously in this scenario, SIP Server incorrectly distributed AttributeCallType=0 (unknown) instead. (ER# 227370882)
SIP Server now correctly processes the following scenario:
INVITE message from an originating party to a destination.18X message containing SDP.PRACK message to the destination and propagates the 18X message containing SDP to the originating party.PRACK, the destination immediately responds with a 200 OK message to the PRACK and with a 200 OK message to the INVITE, which contains the same SDP as the 18X message sent earlier.
200 OK message to the originating party, and the call was not established properly. This issue occurred if the sip-enable-100rel configuration option was set to true. (ER# 232264911, 239431085, 229391998, 237065491)
SIP Server now correctly obtains the value for the WrapUpTime parameter from user data for the consultation call, when setting the agent After Call Work time. Previously, SIP Server incorrectly assigned the WrapUpTime value with user data from the main call instead. This situation occurred when the following options were configured:
user-data-from=consult-user-dataconsult-user-data=inheritedmerged-user-data=merged-onlySIP Server no longer starts a new prompt if the previous prompt ended while call routing was in progress. Previously in this scenario, SIP Server could incorrectly start a new prompt, causing it to later abandon the call due to race conditions. (ER# 235614363)
SIP Server now distributes EventDnOutOfService in the following cases:
REGISTERED.contact value of *.
SIP Server now correctly establishes a media connection on outbound calls made through a Trunk DN configured with reuse-sdp-on-reinvite set to true. Previously, SIP Server failed to establish the connection in the following scenario:
TMakeCall triggers the call to the external destination.INVITE with no SDP to the originator and receives 200 OK with SDP in return.INVITE with SDP is sent to the external destination.180 Ringing with no SDP.183 Session Progress with SDP.183 response to the call originator.200 OK response. ACK request. It completes the re-INVITE transaction on the origination side and establishes the media channel. (ER# 236345041)
SIP Server now accepts re-routed calls in cases where the originating T-Server does not release calls within 500 ms. Previously, calls re-routed back to SIP Server failed in this scenario. SIP Server now waits the length of time specified by the timeout option (extrouter section) before releasing a call. (ER# 239072665)
SIP Server now accepts T-Library client registration requests on a DN with the name exr, even if this device is not configured in the Configuration Layer. Previously, SIP Server rejected such registration requests from the T-Library client. (ER# 241528277)
SIP Server now correctly processes SIP dialogs in cases where, during an INVITE transaction initiated by the session timer, SIP Server receives a 491 Request Pending response. Previously in this scenario, SIP Server inadvertently released the call. (ER# 241528234)
SIP Server now immediately releases a call during the time interval between the moment the call is abandoned on a routing point, but before the 18x response from the target user agent is received. Previously in this scenario, the call was released only after the after-routing-timeout expired and the target user agent was falsely reported as out of service. (ER# 241480853)
In multi-site scenarios, when the REFER method is used to transfer the call to a destination on the same SIP Server, if SIP Server does not receive the INVITE triggered by the REFER within 32 seconds, the corresponding party and subsequently the call itself will be released, preventing the call from getting stuck in SIP Server memory. (ER# 239101871)
SIP Server now rejects RequestSingleStepTransfer requests for a particular call if another call is being held on the same user agent. Previously, there was a limitation for this scenario. (ER# 241061740)
When SIP Server receives a SIP message by UDP, and this message is truncated, it sends a 413 (Request Entity Too Large) response. SIP Server detects the truncated message if the length of the message body is smaller than its Content-Length header.
SIP Server is unable to generate the correct response if the SIP message does not have a header, because it will not have the required call information. This is in accordance with RFC 3261, which states:
The server is refusing to process a request because the request entity-body is larger than the server is willing or able to process. The server MAY close the connection to prevent the client from continuing the request.
If a
413 (Request Entity Too Large) response is received, the request contained a body that was longer than the UAS was willing to accept. If possible, the UAC SHOULD retry the request, either omitting the body or using one of a smaller length.
(ER# 226776378)
SIP Server no longer incorrectly attempts to re-initiate a greeting on a call, instead of connecting the caller to the agent, in rare cases when the greeting completes at the same moment that a requested operation is being performed on that call. (ER# 238930482)
SIP Server now correctly disconnects the only party remaining on the call in scenarios where the two-party call participant's user agent failed to respond to a request within the SIP transaction timeout period (32 seconds). Previously, SIP Server did not disconnect the remaining party, or report the DN corresponding to the failed party as out of service. (ER# 238054521)
SIP Server now selects the correct alternate treatment device when recovering a treatment. Previously, SIP Server sometimes selected the wrong treatment device in the following scenario:
Announcement is being applied to the call.SIP Server now processes all late CANCEL requests in accordance with RFC 3261. Previously in some scenarios, SIP Server failed to send a response to a CANCEL request for a transaction for which a final response had already been sent, resulting in the possibility of stuck calls. (ER# 239535671)
SIP Server now correctly processes dynamic changes to the capacity option for a device. (ER# 239958246)
SIP Server no longer encounters the stuck call situation described in the following scenario:
RequestInitiateTransfer or RequestInitiateConference is issued.RequestReleaseCall, but before the EventDialing was reported on the consultation call.A call party can now see "pushed" video when the AgentVideo parameter in the Extensions attribute is set to from-third-party. (ER# 171694535)
SIP Server no longer sends INFO messages to an endpoint if, on establishing the session with the endpoint, the Allow header of the SIP message did not include INFO. Previously, SIP Server could send INFO messages even if the endpoint did not allow it. (ER# 239296613)
SIP Server now correctly appends the header P-gcti-connid to INVITE messages related to a TRouteCall that contains the 'SIP_HEADERS'='P-gcti-connid' parameter. Previously, SIP Server might have omitted this header from the new INVITE after a call redirect, if the original INVITE corresponding to the TRouteCall was answered with a 302 response. (ER# 205584456)
SIP Server now properly generates the EventReleased in scenarios where SIP Server terminates a call with two participants using TReleaseCall, and one of the SIP endpoints responds with a 481 error, while the other does not respond at all. Previously, SIP Server might have generated the EventReleased without the required AttributeReferenceID. (ER# 164804612)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 handles a race-condition scenario and connects the call with the answered party if the no-answer timeout expires while SIP Server was processing a TAnswerCall request. Previously in this scenario, SIP Server incorrectly released the call (or redirected the call to a no-answer-overflow DN) when the no-answer timeout expired. (ER# 263166651)
SIP Server now correctly completes the Out Of Signaling Path (OOSP) transfer using the REFER method with Replaces. Previously in this scenario, SIP Server sometimes went into an infinite loop that might have led to excessive memory utilization. This issue occurred if the Refer-To header of the REFER method contained the hnv-unreserved character. (ER# 270560291)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 successfully delivers calls to a Voice Treatment Port DN in cases where the parameter Switch-specific Type on the Advanced tab of this DN was set to 0. Previously, SIP Server was unable to deliver calls in this scenario. (ER# 256295773)
SIP Server now correctly executes TRetrieveCall requests when the SIP dialog to a destination is in an early state. Previously, TRetrieveCall requests in this scenario resulted in an EventError. Now, SIP Server correctly handles this scenario if the option sip-early-dialog-mode is set to 1 on the Trunk DN used to reach the destination. Note that the SIP dialog between SIP Server and the call origination side must be in an established state. (ER# 255744048)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 releases the external party from a call that has invoked a transfer using the REFER method. Previously, SIP Server did not release this party from the call, which affected subsequent call flow. In particular, SIP Server reported incorrect values for the AttributeCallState and AttributeThisDNRole parameters in EventEstablished. (ER# 248411009)
SIP Server now properly collects all digits while processing the interruptible treatment PlayAnnouncementAndDigits even if a SendDTMF operation is performed by several TSendDTMF requests. Previously, SIP Server sometimes collected only the first digit. This issue occurred when SIP Server was configured with the option sip-dtmf-send-rtp set to true. (ER# 245982711)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 processes SDPs with the session name, which is passed in the s line, set to a single space character. Previously, SIP Server produced an invalid s line which might have caused a call disconnection and broken audio path during the SIP negotiation session between endpoints. (ER# 246514288)
SIP Server now correctly resolves an IP address from the DN contact when Out-Of-Service check is configured for this DN. Previously in this configuration, SIP Server logged the following error message: Can not resolve hostname xxx.xxx.xxx.xxx, where xxx.xxx.xxx.xxx was the valid IP address. (ER# 321306020)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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:
For SIP dialogs that were started before an High-Availability (HA) switchover, SIP Server now includes the correct value for the received parameter in the Via header of the SIP response. (ER# 240934494)
In 1pcc single-step transfer scenarios using the REFER method, SIP Server now correctly creates a request URI of the outgoing INVITE message by adding a destination host IP to this URI. Previously, SIP Server inserted its own IP-address in the request URI. (ER# 239750207)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 always distributes an EventAttachedDataChanges event at the moment GVP sends a BYE message with data in the body after a voice treatment. Previously, SIP Server skipped this action if the BYE was sent using TCP. (ER# 227897331)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 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 processes 491 Request Pending responses received for INVITE requests. SIP Server handles a 491 response to a particular INVITE by starting a timer, and then re-trying the same INVITE request when the timer expires. Previously, SIP Server was unable to correctly process these 491 requests. (ER# 239101613)
SIP Server no longer tries to restart music treatments that ended after the timeout period specified by the DURATION treatment parameter has expired. Previously, SIP Server would erroneously try to restart the finished treatment. (ER# 237294822)
SIP Server now issues the correct error code in response to 3pcc requests on calls that are disconnected before the request is completed. Previously, if a 3pcc action was requested on a call (for example, RequestHoldCall), but the call was released before the 3pcc action was completed, SIP Server responded to the 3pcc request with the generic error code 100 (Unknown cause). Now, SIP Server responds to the uncompleted 3pcc request with error code 237 (Call has disconnected). (ER# 235342470)
SIP Server now fully synchronizes SUBSCRIPTION requests from RTC-based User Agents (UA) to the backup SIP Server instance. Previously, a synchronization failure could prevent the subscription from working properly after a switchover. (ER# 230813810)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 8.0.001.15. 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 SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in this release of SIP Server.
Support for Network Attended Transfer/Conference (NAT/C). SIP Server now supports the NAT/C feature, which allows agents working in multi-site contact centers to consult with each other before making call transfers or conferences. (ER# 216471926)
Single host deployment. SIP Server now supports deploying both primary and backup high-availability (HA) SIP Server applications, as well as the Stream Manager application, on the same physical host. (ER# 232381886)
Support for AIX 6.1. SIP Server now supports deployments on the IBM AIX Version 6.1 operating system. (ER# 232382102)
Call supervision for transferred calls. SIP Server now supports supervisor monitoring for DNs that are moved from a consultation call to the main call as a result of a two-step transfer.
Note: This is a restricted feature. If you need to enable this feature, please contact your Technical Support representative for advice. (ER 216837596)Enhanced logging for unavailable trunks. SIP Server now issues log event messages to inform users about the availability of the gateway during outbound calls. (ER# 232382020)
Alternate routing for unresponsive DNs. SIP Server now supports
alternate routing for new calls to Genesys SIP Endpoints that fail to
respond to the INVITE request. (ER# 234064653)
Enhanced active out-of-service-detection. SIP Server now provides the
ability to configure the value of the Max-forwards header used
in the OPTIONS messages that SIP Server sends to check the
availability of a particular SIP device. (ER# 232381956)
Support for use of the F5 Networks® BIG-IP® Local Traffic Manager™ to manage the virtual IP address of a SIP Server HA pair (tested on BIG-IP 9.4.7 Build 320.1 Final). For more information about configuring SIP Server for high availability using the F5 BIG-IP Local Traffic Manager, refer to the Framework 8.0 SIP Server Integration Reference Manual. (ER# 232382122)
Dial Plan. SIP Server now supports dial-plan, a new service-type available for Voice over IP Service DNs. Dial-plan DNs are used to create a series of dialing rules that SIP Server applies to the dialed digits that it receives from an endpoint.
INFO messages. SIP Server now
includes a new option, info-pass-through, that lets you control whether SIP Server will pass SIP INFO messages to
a remote device.(ER# 232382152)
Additional support for Outbound IP. SIP Server now includes the following new features to support Outbound IP 8.0:
partition-id, used to select a service
based on the partition to which it belongs.cpd-capability, used to select an outbound
Trunk DN based on the requested call progress detection
(CPD) capability. (ER# 226657941, 226657938)
subscription-id option.
(ER# 227752632)
Configurable beep timer. SIP Server now supports a timer to control the duration of the beep tone used to notify agents participating in an outbound campaign when they are about to be connected to a customer (this option is only available if the outbound campaign is running in the Active Switching Matrix (ASM) mode). (ER# 226657894)
Updated timeout for CPD INFO messages.
Additional Extensions for
TMakePredictiveCall requests. SIP Server now supports two
new key-value pairs, AnsMachine and FaxDest,
that you can include in the Extensions attribute of
TMakePredictiveCall requests.
New media server alarms. SIP Server supports two new alarms used to report the state of the SIP Server subscription to the GVP Resource Manager, for media server functionality.
Updated timeout period for predictive calls on a Routing
Point DN.
INVITE requests) to specific values in the
AttributeCallState included in the TEvent
response to the TMakePredictiveCall request. This release also includes the following corrections and modifications:
SIP Server now correctly processes NOTIFY requests for the Busy Lamp Field (BLF) feature in cases where the entity attribute in the request includes the prefix sip:. Previously, SIP Server ignored these NOTIFY requests. (ER# 233939698)
SIP Server no longer generates the following confusing message to the log output:
TRNFACTORY: WARNING - failed to dispatch ACK message
To avoid confusion, SIP Server now generates the following message instead:
TRNFACTORY: Non matched ACK message: ignored
(ER# 234581588)
SIP Server now properly processes TMakeCall requests on behalf of a DN in the following scenario:
TMakeCall request.EventError message. This issue occurred on DNs configured in Configuration Manager with the option dual-dialog-enabled set to false. (ER# 227384518)
SIP Server now behaves correctly when unable to refresh a session with an agent that is currently being monitored by a supervisor. (ER# 237578791)
SIP Server no longer applies Class of Service (COS) outbound dialing rules to calls made
to a Routing Point DN. Previously, when a call was made from a DN to a Routing Point and there was a COS configured and assigned to this DN, SIP Server might have incorrectly applied,
for example, digit manipulation based on out-rules to the dialed digits.
(ER# 229945825)
SIP Server now uses the correct key USER_ANN_ID in the Extensions attribute of the TreatmentEnd event
that it sends in response to a RecordUserAnnoucement treatment
request. Previously, SIP Server used the incorrect key name
USER_ANNC_ID instead of USER_ANN_ID.
(ER# 176754898)
SIP Server now classifies internal calls made to an agent as
Private by default. Previously, SIP Server classified calls
from internal DNs as having the type Unknown unless the option
internal-bsns-calls was enabled, in which case SIP Server
would classify them as Business-type calls. This effectively disabled
no-answer supervision for these calls, even if the nas-private
option was set to true.
Now, by classifying these calls as Private by default, you
can successfully use the nas-private option to enable
no-answer supervision for internal calls.
When processing call routing, SIP Server now correctly distributes an
EventReleased message to an endpoint that responds to an
incoming INVITE message with a 480 Temporarily Unavailable.
Previously, SIP Server distributed an EventAbandoned message
regarding this DN, which cleared the call from memory in the Universal
Routing Server (URS). This error occurred when SIP Server was configured
with the option divert-on-ringing set to false
and event-ringing-on-100trying set to true.
(ER# 230701125)
When SIP Server receives a Register request with the values
Contact: * and Expires=0, it now checks that the
Address of Record (AOR) in the Request-URI is valid, and that
the Request-URI belongs to the internal domain managed by SIP Server. If
the AOR and domain are correct. SIP Server tries to remove all the contacts
for the DN provided in the user part of the TO address. Even
though no contact is currently registered with SIP Server for that DN, SIP
Server sends a 200 OK instead of a 403 Forbidden.
Previously in this scenario, SIP Server sometimes failed to recognize the
asterisk (*) as the Contact value, causing unregister requests from the SIP
phone to return a 403 Forbidden instead of the 200
OK it required. (ER# 230988644)
SIP Server now correctly includes the SDP in the refresh
re-INVITE message that it sends after a failover in high
availability configurations. Previously, SIP Server sometimes failed to
provide the SDP in SIP messages after a failover from backup to primary SIP
Server, resulting in a dropped dialog by the client. (ER# 233939671)
On receiving a 488 (Not Available Here) in response
to a re-INVITE request, SIP Server now forwards this response on to the other call leg, allowing the User Agent to retry the
re-INVITE with different session parameters. Previously, SIP
Server did not forward the 488 (Not Available Here) response to the other call leg. (ER#
220716681)
SIP Server now correctly generates a 403 Forbidden in response to INVITE requests that it sends to
Extension DNs with no agents logged in. Previously, SIP Server
responded incorrectly with a 481 (Call Leg/Transaction Does Not
Exist). This issue only occurred if the agent DN was configured with the option
reject-call-notready set to true.
(ER# 219864653)
SIP Server now distributes only one call per agent that has logged in to an ACD queue. Previously, SIP Server was able to distribute two calls to the same agent if the agent returned to the queue after the previous call released, and if there were two or more calls waiting in the queue for service. (ER# 233711245, 226311308)
SIP Server now correctly handles monitoring sessions in the following scenario:
RequestMonitorNextCall for a call
already established between the caller and an agent.180 Ringing message.sip-ring-tone-mode in the SIP Server
application is set to 1.SIP Server now responds to INVITE requests that target DNs
configured with an asterisk (*) in the contact option by
correctly issuing a 404 Not Found message. Previously, SIP
Server mistakenly responded with a 481 Call/leg Does Not Exist
message. (ER# 230140943)
SIP Server now interprets an empty <activities>
element in the RPID (Rich Presence Extensions to the Presence Information
Data Format (PIDF), as described in RFC 4480) the same way that is
interprets the absence of the <activities> element. On
receiving a NOTIFY message containing an empty
<activities> element in the RPID, if the agent is in the
NotReady state, SIP Server generates an
EventAgentReady message. (ER# 217381160)
SIP Server now supports appending the version=2.0 field in
the Content-Type header of SIP NOTIFY messages, as
follows:
Content-Type: message/sipfrag;version=2.0
In accordance with RFC 3420, this version field is included
as an optional parameter in the message/sipfrag header.
(ER# 222235466)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.16. 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 SIP Server.
This is a hot fix for this product. This section describes new features that were introduced in this release of SIP Server.
contacts-backup, used to configure alternative URI addresses for supported DNs. This option is currently available for Trunk DNs and Voice over IP Service DNs with service-type set to softswitch.contact option cannot be reached.Trunk DN used to control presence subscription, in cases where more than one Cisco SIP trunk is deployed.
This release includes the following corrections and modifications:
SIP Server now supports setting the geo-location option on Routing Point DNs. If you define the option on a Routing Point, the value overrides the geo-location already set for the call. However, if the Extensions attribute in a TRouteCall request includes a geo-location value, then this value is given the higher priority.
The geo-location on a Routing Point configuration supports the following scenarios:
TRouteCall is submitted on the Routing Point. If TRouteCall does not have a geo-location value in the Extensions attribute, then the DN-level parameter is used to select a routing destination.
router-timeout expires when TRouteCall is in progress. The default routing destination is selected based on the default-dn and geo-location options as configured on the Routing Point.
(ER# 238681111)
SIP Server now correctly ignores NOTIFY messages with an open state, in cases where the message is sent from Cisco CallManager (CCM) for an agent that was previously logged out from its corresponding DN by a T-Library client. Previously, SIP Server incorrectly processed NOTIFY messages in this scenario, mistakenly setting the agent to a logged in state. (ER# 238706759)
SIP Server no longer sets a DN to the OutOfService state in cases where a re-INVITE, triggered by the session-timer, results in a 481 Call/Transaction Does Not Exist error. (ER# 238722463)
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 7.6.008.15. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
Note: This release incorporates several Corrections and Modifications from the 7.6.x version of SIP Server, which were made after the initial 8.0.x release. For information about these fixed ERs, see the entries for release 7.6.000.76 and higher in the SIP Server 7.6 Release Note.
This release also includes the following corrections and modifications:
In high availability (HA) configurations, SIP Server now includes the
correct Session Description Protocol (SDP) in the refresh
re-INVITE message that it sends after a switchover from primary to
backup SIP Server. Previously in this scenario, SIP Server sometimes sent the
re-INVITE with no SDP, causing the client to drop the dialog.
(ER# 233939671)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.15. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
Note: This release incorporates several Corrections and Modifications from the 7.6.x version of SIP Server, which were made after the initial 8.0.x release. For information about these fixed ERs, see the entries for release 7.6.000.76 and higher in the SIP Server 7.6 Release Note.
This release also includes the following corrections and modifications:
SIP Server has been rebuilt to correct a minor build 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 7.6.008.15. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
Note: This release incorporates several Corrections and Modifications from the 7.6.x version of SIP Server, which were made after the initial 8.0.x release. For information about these fixed ERs, see the entries for release 7.6.000.76 and higher in the SIP Server 7.6 Release Note.
This release also includes the following corrections and modifications:
SIP Server now correctly processes TRetrieveCall requests when
the DNs for both call participants are configured with the option
reuse-sdp-on-reinvite set to true. Previously in this
case, the TRetrieveCall requests could sometimes fail.
(ER# 231207441)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.15. 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 SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in this release of SIP Server.
Media Server Reliability. SIP Server now supports media reliability for conference calls, call recording, and supervisor feature services. (ER# 226236006, 226236008, 226236010)
Playing Multiple Media Files. SIP Server can now play different media files for various contact center music on hold and video on hold requirements, from a single instance of SIP Server. (ER# 226835456)
Contact Header Handling Options. SIP Server supports two methods for
handling the Contact header in REGISTER requests.
It can preserve information contained in the header, or replace the
user-name part of the URI with the DN name. (ER# 224356760)
Recording for consultation calls. SIP Server now supports regular call recording on consultation calls. (ER# 226313380)
Busy tone for more than five seconds. SIP Server can now play a busy tone for five seconds or longer. (ER# 223964491)
Mapping agent actions to T-Library events. SIP Server can now support
agent log in and log out functionality from endpoints using
SUBSCRIBE and NOTIFY requests. (ER# 225459591)
Alternate gateway selection. In cases where SIP Server selects a
Trunk DN that represents a gateway, but all lines for that
gateway are busy, occupied, or otherwise out-of-service, SIP Server will
silently try to reach the destination using another gateway
Trunk DN, if one is available. SIP Server remembers the failed
Trunk DN and, if more than one Trunk DN is
configured, avoids trying it again in the future. (ER# 226255573)
Backwards compatibility for GVP. In standalone deployments with GVP,
SIP Server no longer overwrites the To: header when
redirecting a call to the GVP IPCS resource selected by the GVP Resource
Manager, allowing the IPCS to play the application based on the DNIS. This
restores backwards compatibility supported by SIP Server 7.5. In SIP Server
7.6, SIP Server might have overwritten the original To: header
during the redirect operation. (ER# 226313376)
This release also includes the following corrections and modifications:
SIP Server no longer inadvertently drops the main call when an agent
initiates a TSingleStepConference request and SIP Server receives
a 486 Busy Here message from the media server. (ER# 219139759)
When operating in a high-availability environment, after a switchover, SIP
Server no longer reports a DN configured with the
use-register-for-service-state option set to true as
out of service. Previously, this issue could occur if the DN was in the
in-service state before the backup SIP Server started and no
activity was reported on that DN. (ER# 217632696)
SIP Server now correctly sends an INVITE message to the second
instance of the recorder service if the first instance of the recorder service
fails. Previously, SIP Server was sometimes unable to send this
INVITE, resulting in a call being established without the recorder
service. (ER# 189554684)
When a call is placed to an endpoint configured with the option
dual-dialog-enabled set to false, SIP Server now
always sends new INVITE requests to the endpoint, even if it is
already on another active call. Further SIP Server actions depend on the
response from the endpoint. If the endpoint rejects the new
INVITE, SIP Server generates an EventDestinationBusy
message and sends the rejection response to the originator of the new call. If
the endpoint accepts the new call, SIP Server generates an
EventRinging/EventEstablished message and connects the caller to
the endpoint. This behavior is backwards compatible with SIP Server 7.5 and
7.6. For versions starting with release 8.0.000.12, SIP Server did not send new
INVITE requests to endpoints with this configuration if already on
a call. (ER# 229557255)
SIP Server no longer inadvertently generates
EventDialing/EventDestinationBusy messages for endpoints that were
previously on a call, but later left the call after a single-step transfer.
Previously, SIP Server might have generated these events if, after the
transfer, two external parties were on the call and one of them sent an
INVITE with hold SDP. (ER# 224741580)
When an agent logged in with workmode set to
AfterCallWork completes a transfer, SIP Server now always sends an
EventAgentNotReady message before the EventReleased
message for the main call. This is compatible with the behavior for SIP Server
7.5. For versions 7.6 and 8.0.0.x, SIP Server might have sent the
EventAgentNotReady message after the EventReleased
for the main call. This behavior was possible when the option
inherit-bsns-type was set to true. (ER# 222450846)
SIP Server now generates EventRinging properly in scenarios
where the call is routed and redirected multiple times before delivery to an
agent. (ER# 112279910)
SIP Server no longer inadvertently ends a call by sending BYE
messages to the MCU and other established parties in a conference. Previously,
in scenarios where all internal parties had left the call, SIP Server might
have inadvertently ended the call before a transfer to an external party is
completed. (ER# 120709466)
SIP Server now correctly rejects INVITE requests, sending a
603 Decline message, in cases where the INVITE
attempts to initiate a call to a destination that was set to Do Not
Disturb by the corresponding T-Library request. Now, if the T-Library
request attempts to initiate a call to a device currently in the Do Not
Disturb state, SIP Server rejects the request and generates the error
message Destination is in invalid state. Previously, SIP Server
mistakenly sent a 404 Not Found message in response to
INVITE requests to a destination which was set to Do Not
Disturb state by the T-Library request. (ER# 135378601)
SIP Server now correctly processes INVITE requests where the
From header indicates the Routing Point where the call originally
arrived, instead of the media gateway from which the INVITE was
sent. Previously, SIP Server sometimes created a new party for the call instead
of clearing and transferring the call. (ER# 135559194)
SIP Server now distributes DNBackInService or
DNOutOfService messages whenever a DN becomes available or
unavailable due to a configuration change. These messages are issued in the
following scenarios:
SIP Server now generates an EventAgentNotReady message on
receiving a NOTIFY message that indicates a
Proceeding state, in cases where there is no call (dialed or
established) to the agent. (ER# 177468673)
In high availability (HA) deployments, SIP Server now correctly ends a music or treatment service after a switchover from the primary to the backup SIP Server instance. Previously, SIP Server sometimes continued playing the treatment even after the call was answered, in cases where the switchover occurred while providing the treatment. (ER# 177788136)
SIP Server no longer generates the USER_ANN_ID parameter in
EventTreatmentEnd messages, in cases where recording did not start
for an announcement played as a result of a
TreatmentRecordUserAnnouncement request. Previously, the inclusion
of this parameter indicated that recording was successfully completed, even
though recording had not started. (ER# 190471591)
SIP Server now correctly places the supervisor into a monitoring session when an agent greeting (configured on the Agent Login for the Supervisor) stops playing. Previously, SIP Server placed the supervisor into the conference (not monitoring session) after the greeting stopped playing. (ER# 196058736)
SIP Server now correctly issues an EventError message in
response to a TSingleStepTransfer request, in cases where the
contact option in the Voice over IP Service DN that
represents the MCU is not configured, and no other MCU DN is present in the
configuration. Previously, this scenario sometimes caused SIP Server to stop
responding. (ER# 214543479)
SIP Server now correctly handles the following scenario:
INVITE in time.TReconnectCall or TRetrieveCall requests.INVITE for a consultation
call.SIP Server no longer allows a supervisor to participate in more than one monitoring session at a time. Previously, SIP Server inadvertently allowed more than one monitoring session for an individual supervisor. (ER# 218570309)
SIP Server now generates EventDestinationBusy messages with the
correct CallState parameter:
MakeCall request replies with a
404 NotFound error, SIP Server generates an
EventDestinationBusy with a call state of 11.486 Busy here, SIP Server
generates an EventDestinationBusy with a call state of
6.EventDestinationBusy with a call state of 6.
(ER# 219714837)
SIP Server now places the correct value for the $AGENTDN$ and
$AGENTID$ parameters in the recording file name for outbound
calls. (ER# 223291343)
SIP Server no longer ignores the reuse-sdp-on-reinvite option
when set to true. Previously, SIP Server could send a
re-INVITE message with no SDP to a user agent DN configured with
this option, resulting in the failure of several user agent operations,
including TRetrieveCall request operations. (ER# 223548663)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.15. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release also includes the following corrections and modifications:
When operating in a multi-site environment, different instances of SIP Server can now correctly reconnect with each other or with other T-Servers. Previously, when built with earlier releases of TSCP, SIP Server could not always re-establish lost connections. (ER# 226894601)
SIP Server can now accept TSingleStepTransfer requests from a
monitored party in a conference, when that party is currently set on hold. In
release 7.5, SIP Server rejected such transfer requests, causing subsequent
T-Library requests to also fail. Now, since SIP Server is able to process
TSingleStepTransfer requests from the monitored, on-hold party,
SIP Server no longer has a problem processing subsequent T-Library requests.
(ER# 219739772)
With the Inter Server Call Control/Call Overflow (ISCC/COF) feature enabled,
SIP Server now correctly sends ISCCRequestGetCallInfo requests to
the site of the incoming call only. Previously, SIP Server sent
ISCCRequestGetCallInfo requests to all connected sites, creating
unnecessary traffic overload. (ER# 227333467)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.09. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release also includes the following corrections and modifications:
In a hot standby high availability (HA) configuration, SIP
Server now correctly places an out-of-service Trunk DN back in
service after a failover. Previously, when a Trunk DN representing
an unavailable media gateway was placed in the out-of-service state before a
switchover (Active Out-of-Service detection enabled), and the gateway
subsequently became available again after the switchover, SIP Server was unable
to mark and report the trunk as being back in service. (ER# 226547902)
SIP Server now properly processes the following call flow scenario:
INVITE request with a hold SDP.200 OK message.ACK message, followed immediately
by a new re-INVITE without SDP. At this moment, the
re-INVITE transaction with the other endpoint is still in
progress.INVITE from the gateway with a
200 OK (hold SDP).INVITE transaction with the endpoint is
completed, SIP Server sends re-INVITE requests to both the
gateway and the endpoint, restoring the connection between them.Previously, SIP Server did not complete step 5 of this scenario, which
prevented SIP Server from being able to provide a final response to any
subsequent re-INVITE requests from the gateway.
(ER# 226427672)
SIP Server can now successfully connect an inbound call if an
UPDATE message from the gateway arrives while the call is in a
ringing state. Previously in this situation, SIP Server was sometimes unable to
establish the connection. (ER# 226294795)
SIP Server no longer issues unnecessary Standard log messages
(00-06080) during the startup initialization of the SIP Server.
Previously, SIP Server issued a message stating that the mandatory
configuration option sip-replaces-mode was not
found.
(ER# 221891011)
SIP Server now correctly rejects new re-INVITE messages
(issuing a 500 Error code), in cases where SIP Server receives the
re-INVITE from a particular endpoint before it receives an
ACK message from the same endpoint, in response to a previous
re-INVITE. In this case, SIP Server rejects the new
re-INVITE and maintains the call connection. Previously, SIP
Server inadvertently accepted the new re-INVITE, causing it to
immediately drop the call. (ER# 223113022)
SIP Server now correctly includes the key USER_ANN_ID in the
EventTreatmentEnd message when a
RecordUserAnnouncement treatment is completed. Previously, SIP
Server sometimes did not provide a value for this key. (ER# 223952981)
SIP Server is now able to stop playing
RequestPlayAnnouncementAndDigits treatments with multiple prompt
elements, as soon as the first DTMF digits is collected.
To support this functionality, a new Application-level option,
sip-treatment-dtmf-interruptable has been added.
sip-treatment-dtmf-interruptable
Default Value: false
Valid Values: true, false
When set to true, SIP Server will stop playing all prompt elements
while processing RequestPlayAnnouncementAndDigits treatments, as
soon as the first DTMF digit is collected. When set to false, SIP
Server stops playing the current prompt, but then immediately after digit
collection, starts playing the next prompt.
Previously, SIP Server always played subsequent prompts after digit collection,
as it does now when sip-treatment-dtmf-interruptable is set to
false. (ER# 225378530)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.09. 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 SIP Server.
This is a hot fix for this product. This release does not contain new features or functionality.
This release also includes the following corrections and modifications:
SIP Server now sends CANCEL requests to the address where it
sent the preceding INVITE request. Previously, SIP Server might
have sent the CANCEL request to the address identified in the
Request-Uri parameter of the INVITE request, in which
case the CANCEL request was lost. (ER# 220555361)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release of SIP Server is built with T-Server Common Part (TSCP) release number 7.6.008.09. 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 SIP Server.
There are no restrictions for this release. This section describes new features that were introduced in this release of SIP Server.
Additional support for Polycom firmware. SIP Server now supports Polycom software releases "SIP 3.1.3revC" and "SIP 3.2.2" on all Polycom SoundPoint IP phone models which run these respective versions.
TCompleteTransfer using REFER or
REFER with Replaces. SIP Server now supports the
TCompleteTransfer operation by using either the SIP
REFER method or the SIP REFER method with the
Replaces header based on conditions described below.
TCompleteTransfer using the SIP
REFER method if:Replaces header
in the REFER method.TCompleteTransfer using the SIP
REFER method with Replaces if both the
external transferred party and the external transfer destination party
support the Replaces header in the REFER
method. Notes:
REFER for the
TCompleteTransfer operation for calls with MCU (Multipoint
Conference Unit) involved. For example, a call is monitored or
emergency recording is applied to a call. If regular call recording is
applied to the original call, the REFER method can be used
for the TCompleteTransfer operation.TCompleteTransfer is done using the
REFER method, SIP Server stays in the signaling path. If
TCompleteTransfer is done using the REFER
method with Replaces, SIP Server is taken out of the
signaling path.To support this functionality, the new options
sip-replaces-mode (Application and DN-level) and
transfer-complete-by-refer (DN-level) for DNs of type
Trunk are introduced.
sip-replaces-mode
Default Value: 0
Valid Values: 0, 1, 2
This option specifies the SIP method that will be used by SIP Server to
complete a two-step transfer.
0, SIP Server will use the
REFER method if the
transfer-complete-by-refer option is set to
true.1, SIP Server will use the
REFER method with Replaces if the
Allow header contains REFER as a supported
method and the Supported header contains
Replaces. If REFER with Replaces
is not supported by a device, then TCompleteTransfer will
be done using the REFER method. If a device does not
support the REFER method, then the transfer will be
completed using the re-INVITE method.2, SIP Server will use the
REFER method with Replaces to process
TCompleteTransfer. The Allow and
Supported headers will not be analyzed.Notes:
sip-replaces-mode option is not applicable
to a DN of type Trunk if the
sip-server-inter-trunk option in the DN configuration is
set to true.refer-enabled option
must be set to true in the DN from which a call party is
transferred to another destination during a two-step transfer.transfer-complete-by-refer
Default Value: false
Valid Values: true, false
When set to true, this option enables SIP Server to
complete a two-step transfer by sending a REFER message to the
party in the primary call. SIP Server uses the same content as in the
REFER message sent for a single-step transfer. For this option
to work, you must set refer-enabled on the Trunk
DN to true.
Limitations for this option include:
REFER is not used if the primary party on the
consultation call is involved in a conference.REFER is not used if the call is currently being
recorded.Trunk DNs configured
between different SIP Server instances, and is ignored on
Trunk DNs where the sip-inter-trunk option is
set to true.(ER# 216471942)
Automatic Agent Logout. SIP Server can now automatically log out an
agent after a specified period of inactivity, ensuring accurate reporting
of agent activity. Automatic agent logout can be configured for agents in a
NotReady status, or more strictly for agents in either a
NotReady or Ready status in a work-related mode
(for example, AfterCallWork).
Agent activity is determined by monitoring the following:
To support this feature, two new options have been introduced:
auto-logout-timeout and auto-logout-ready.
auto-logout-timeout
Default Value: 0
Valid Values: 0, or any positive integer up to 35791
Changes Take Effect: Immediately
Enables automatic agent logout and specifies the length of time after which
the logout occurs (in minutes). To enable this feature, enter a value of
1 or greater; the agent is allowed to remain inactive for this
length of time before they are automatically logged out. To disable this
feature, enter a value of 0 (default).
You can configure this option in the TServer section of the
following objects (listed in order of precedence):
Agent Login object (Annex tab)Annex tab of the ACD Position or
Extension DN) that represents the device where the agent
is logged on.Annex tab of the Routing Point
or ACD Queue DN) that represents the queue where the agent
is logged on. Application object (Options
tab), which specifies the server-wide default. auto-logout-ready
Default Value: false
Valid Values: false, true
Changes Take Effect: Immediately
Enables a stricter enforcement of the automatic agent logout policy (as set
in the related auto-logout-timeout option). If set to
true, SIP Server will log out the agent regardless of agent
state. If set to false, SIP Server will not log out agents
when in the following agent states: Ready,
NotReady/ACW, NotReady/AuxWork,
NotReady/LegalGuard.
You can configure this option in the TServer section of the
following objects (listed in order of precedence):
Agent Login object (Annex tab)Annex tab of the ACD Position or
Extension DN) that represents the device where the agent
is logged on.Annex tab of the Routing Point
or ACD Queue DN) that represents the queue where the agent
is logged on. Application object (Options
tab), which specifies the server-wide default. (ER# 216471920)
Enhanced geo-location support. SIP Server now supports the ability to
assign the geo-location for a call from the routing strategy,
which takes precedence over geo-location configured at the
DN-level in Configuration Manager. If configured for it, the routing
strategy can ensure that the TRouteCall or
TApplyTreatment request contains the attribute
Extensions with the key name geo-location, whose
value the SIP Server then uses as the preferred geo-location
for the current call. When selecting a particular SIP resource, SIP Server
will first consider those resource DNs whose configured
geo-location matches the preferred geo-location
assigned for the call.
If the routing strategy sends a TRouteCall or
TApplyTreatment request that does not include the
geo-location parameter, SIP Server will de-activate the
preferred geo-location mode for the call, and use the regular
selection procedure instead (typically the round-robin method for load
distribution). (ER# 216970196)
Support for TClearCall requests. TClearCall
instructs SIP Server to delete all parties from any type of call. This
includes call operations currently underway or currently queued (for
example, a HoldCall operation delayed while waiting for a
response to an INVITE request). For any pending request that
is canceled by the TClearCall operation, SIP Server generates
an event error with code TERR_DISCON_CALL(237).
(ER# 216608907)
SIP Server now synchronizes the SIP registration
contact header for a particular device across both primary
and backup SIP Server instances. The primary SIP Server sends the
contact information directly to the backup SIP Server using the HA
link, as well as through the Configuration Server as in previous
versions of SIP Server. (ER# 216471957)
Enhanced reliability for media services. SIP Server now immediately
re-invites the next available media server, in cases where a
BYE message is received from the media server while playing
music-on-hold or ringback tone, or when the HTTP stream is disconnected.
SIP Server selects the next available media service (for example, an
instance of Stream Manager) in a round-robin fashion. If a service fails to
start at a particular instance of Stream Manager, SIP Server tries the next
instance, but if the service cannot start on any of the instances, the call
fails permanently. To avoid looping or network overload, SIP Server does
not retry any Stream Manager instances where the service had previously
failed. (ER# 216471951, 221108036)
Nailed-Up Agent. SIP Server can now provide a persistent SIP session for agents that require a dedicated connection to the contact center—typically for TDM agents dialing in to the contact center from the PSTN. The session can be released only after the agent hangs up the phone (1pcc release call).
To support this feature, the new option line-type has been
added to the agent DNs (Annex tab of ACD Position
or Extension DNs):
00, 10) or a nailed-up line (1). If set to
1, when a call to this DN is released due to a 3pcc request
(TReleaseCall, SingleStepTransferCall, or
CompleteTransfer), SIP Server does not end the SIP session
with this DN. Instead, SIP Server parks the nailed-up line on the
gcti::park device, where the SIP session is maintained and the
DN is able to make 3pcc calls or receive new calls. This behavior is
typically required for TDM DNs behind a media gateway, where the agent
requires a dedicated connection to the contact center for the duration of a
work session.
Note: Nailed-up DNs must not be configured with the
sip-cti-control option (talk, hold).
In addition, for each nailed-up DN you must also configure the following options:
refer-enabled to false.dual-dialog-enabled to false.reject-call-notready to true
(recommended, not
mandatory).(ER# 216471954)
Enhanced Agent No-Answer Supervision. SIP Server can now apply Agent
No-Answer Supervision functionality (alternate routing or forced agent
logout) in cases where an agent DN returns a 4xx rejection
response to an INVITE request. For example, the agent pressed
Do Not Disturb on their handset, and the agent DN returns a
486 Busy here message in response to an INVITE
request. Previously in this scenario, SIP Server was unable to place the
agent DN in NotReady status, and the call was passed
continuously between the queue and the agent endpoint.
set-notready-on-busy.set-notready-on-busy
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When this option is set to true, SIP Server places an agent in
the Not Ready state (an EventAgentNotReady
message is distributed) if only one call is distributed to the agent (that
is, the agent was not previously engaged in a call) and his or her endpoint
responds to the INVITE with a 4xx,
5xx, or 6xx message. In addition, a
ReasonCode key with a value equal to a returned error will be
reported in the Extensions attribute in the
EventAgentNotReady message. If a call is distributed to an
agent via an ACD queue, the agent is placed in the Not Ready
state and the call is diverted to the same ACD queue (at the end of the
queue).
(ER# 214217938, 216471917)
Support for Quality of Service (QoS). SIP Server can now set QoS bits to a user-defined value in order to prioritize SIP signaling traffic.
To support this functionality, the new Application-level option
sip-ip-tos has been introduced.
sip-ip-tos
Default Value: 0
Valid Values: Integer 0-255, in decimal format or in hexadecimal format
with 0x prefix
Changes Take Effect: Immediately
Defines the value of the Type of Service (TOS) byte in the IP header of SIP messages sent by SIP Server. Depending on the network configuration, the TOS byte is treated as either:
For example, the following values may be used to assign a higher priority to SIP packets:
Notes:
On most operating systems, applications running on behalf of non-privileged
user accounts are not permitted to set a non-zero TOS value, so you may
have to perform additional actions to enable this functionality. In
particular:
CAP_NET_ADMIN
capability (that is, run from the root account). HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\DisableUserTOSSetting
= (DWORD) 0(ER# 216608913)
Stuck Calls Cleanup. SIP Server now supports the detection and cleanup
of stuck calls. The call-cleanup section on the
Options tab of the SIP Server Application object
contains configuration options that are used to control detection and
cleanup of stuck calls in SIP Server. For more information, see the Call
Cleanup Section in the Framework 8.0 SIP Server Deployment Guide.
(ER# 216471948)
Support for early media. SIP Server now supports the exchange of early media before a particular session is accepted (for example, to provide an audio treatment before the call is answered, thereby avoiding toll charges for the caller).
SIP Server provides support for early media using an offer/answer
exchange of provisional responses and UPDATE requests, in
order to manage the session parameters (SDP) required to deliver the early
media. All early media dialog activity takes place before the 200
Ok response, at which time the dialog is no longer provisional, but
confirmed.
Support for the UPDATE method as controlled by the
sip-early-dialog-mode option has been introduced.
00, 1For devices that support an offer/answer exchange using the
UPDATE method, SIP Server will send the UPDATE to
the called device if the Trunk DN is configured with
sip-early-dialog-mode set to 1. If set to
0, this functionality is disabled.
The following scenarios are supported:
SendDTMF, Hold, or Retrieve on
the calling party.For compatibility reasons, DNs with the option
sip-server-inter-trunk set to 1 also support
early media, if support for early media (PRACK) is reported in
the Allow header.
Support for cost-free early media as controlled by the
charge-type option has been introduced.
0, Effect varies according to DN type| DN type | Default Effect |
|---|---|
| Voice over IP Service | free |
| Trunk Group | charged |
| Extension | charged |
0, 1 Specifies whether a charge will be incurred for services supplied by this DN.
Note: This option is currently supported on Trunk
Group, Extension, and Voice over IP
Service DNs only.
Note: Once the call is established (200 Ok is sent),
no further early toll-free services are possible.
In addition to the sip-early-dialog-mode and
charge-type options, Genesys recommends that you also set the
ringing-on-route-point option on the SIP Server
Application object to false.
Support for ITU-T Recommendation E.164. SIP Server now supports the E.164 recommendation for the international public telecommunication numbering plan, as required to achieve SIP Connect certification. (ER# 216608934)
Support for Paraxip Media Gateway. This release of SIP Server supports Paraxip Media Gateway software version 2.2.3.
This release also includes the following corrections and modifications:
SIP Server no longer has memory leaks when calls are abandoned on a Routing Point. Previously, SIP Server memory utilization increased in this scenario. (ER# 187111388)
SIP Server now always propagates the AttributeReferenceID and
AttributeReason parameters obtained from the
RequestSetAgentNotReady message into the corresponding
EventAgentNotReady message. Previously, SIP Server could sometimes
miss forwarding these attributes when invoking a RequestMakeCall
towards a particular DN. (ER# 219261461)
SIP Server now associates the correct Trunk DN configuration
object with the inbound call leg. Previously, SIP Server sometimes chose the
wrong Trunk DN. As a result, if a high availability switchover
occurred after the call started but before call routing took place, SIP Server
might have used the wrong SIP method to route the call. (ER# 223309254)
SIP Server now adds two new header fields, Route and
Max-Forwards, when sending PRACK requests to confirm
a reliable provisional response. Previously, SIP Server did not include these
headers in the PRACK request. (ER# 177953290, 177787885)
In multi-site environments, SIP Server now correctly handles concurrent
INVITE transactions that result from scenarios where a
CollectDigits treatment is applied to a Routing Point located on
site 1 and the call originator is located at site 2. Previously in this
scenario, SIP Server could not complete the INVITE transaction
initiated towards the media gateway, resulting in silence for the call
originator. (ER# 177787905)
SIP Server now correctly releases canceled outbound calls after a time
interval specified by the cleanup-idle-tout option. Previously,
when an agent endpoint initiated a 1pcc outbound call, then subsequently
dropped the call after receiving a provisional response, this resulted in a
stuck call. (ER# 215129916)
SIP Server now correctly responds to an incoming INVITE request
with a 404 Not Found message if the inbound call is placed to a
non-existent Routing Point DN, where the DN is deleted from
Configuration Manager while in a disabled state. Previously, SIP Server
attempted to process the call to the deleted DN, resulting in inconsistent and
incorrect behavior. (ER# 184633165)
SIP Server now responds to incoming SIP REGISTER requests with
a 404 Not Found message, where the registering DN is deleted or
disabled in Configuration Manager. Previously, SIP Server incorrectly accepted
the REGISTER request. (ER# 173677111 )
SIP Server now correctly processes provisional responses from media gateways
containing early media SDPs (such as 183 Session Progress
messages). Previously in some scenarios, this issue caused an invalid SDP
handshake sequence, and an audio path could not be established. (ER# 214755172)
SIP Server now supports recovery-timeout functionality for the following
types of DNs: Extension, ACD Position, and
Voice Treatment Port. SIP Server is now able to place these DNs
back in service after a configured recovery-timeout period
expires, when the DN is placed out of service for not responding to the
INVITE. Previously, these DNs remained out of service even after
the recovery-timeout expired. (ER# 211767461)
SIP Server can now correctly select a Trunk DN for consultation
calls by geographic location. Previously, SIP Server sometimes selected a trunk
with the same geographical location as the DN that originated the consultation
call. As a result, SIP Server might have selected a trunk at the same
geographic location as the trunk for the primary call. (ER# 38947944)
SIP Server no longer sends duplicate P-Charging-Vector headers
in the outgoing INVITE message when routing a call to a target
destination. (ER# 216974430)
A new valid value * (asterisk) for the contact
option at the DN-level is introduced. If SIP Server receives a
REGISTER message with expires=0, it marks the contact
for the DN with the value *. Any attempt to make a call to/from a
DN with contact set to * will fail, and an
EventError message will be issued. (ER# 198992680)
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.
The Framework 8.0 SIP Server Deployment Guide contains incomplete information about the option make-call-alert-info. This option also applies to TInitiateTransfer and TInitiateConference requests, in addition to TMakeCall. The next published version of the document will include the following updated option description.
make-call-alert-info
Default Value: No default value
Valid Value: Any string
Changes Take Effect: Immediately
The contents of this field are passed in the Alert-Info header of the INVITE message sent to the origination party in response to any of the following requests:
<file://Bellcore-dr3> turns on a triple ring on Cisco 7940 endpoints.
(ER# 267711173)
The Framework 8.0 SIP Server Deployment Guide contains incomplete information about the option contacts-backup. This option can only be applied to Trunk and Voice over IP Service DNs, and also requires that you enable Active Out-of-Service Detection. The next published version of the document will include the following updated option description.
contacts-backup
Default Value: No default value
Valid Value: A comma-separated list of any valid SIP URI
Specifies a list of alternative SIP URI addresses to be used in cases where the SIP URI specified in the contact option cannot be reached. You can apply this option to Trunk and Voice over IP Service DNs only. SIP Server uses the oos-check options (oos-check, oos-force, recovery-timeout) to determine which node in the cluster is currently available to handle SIP requests. Configure each URI using the following format:
[sip:][number@]hostport[;transport=(tcp/udp)]
Note: The same combination of IP/hostname, port, and protocol must not be used in more than one DN.
For integration with Cisco CallManager (CCM), you must configure this option on the Trunk DN used to control presence subscription, in cases where more than one Cisco SIP trunk is deployed.
This section provides the latest information on known issues and recommendations associated with this product.
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:� |
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: 8.0.400.90 |
In Out Of Signaling Path (OOSP) scenarios where the call goes through several Trunk devices, in order for SIP Server to control (or filter) the mapping of custom SIP headers from TRouteCall and/or TMakePredictiveCall requests to an outgoing REFER request, the enable-extension-headers configuration option should be specified on the referred-by Trunk device. (ER# 268242959)
| Found In: 8.0.400.52 | Fixed In:� |
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 |
Prior to release 8.0.300.40 of SIP Server, the maximum valid value for the for the option subscribe-presence-expire was 3600. Starting in release 8.0.300.40 and for later releases, this maximum value is increased to 259200. Note that the Framework 8.0 SIP Server Deployment Guide only states the valid values of 10 to 259200 seconds, as supported by the latest release of SIP Server. (ER# 252785342)
| Found In: | Fixed In:� |
The Network Asserted Identity mechanism is not supported on any scenarios involving Routing Points except predictive calls. (ER# 263094169)
| Found In: 8.0.400.42 | Fixed In:� |
| 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:� |
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 |
If a supervisor connects to a multi-site call, the EventPartyAdded message is not delivered to the remote SIP Servers. Only T-Library clients registered on DNs configured on the same SIP Server as the Supervisor will receive the EventPartyAdded message. (ER# 256032291)
| Found In: 8.0.400.25 | Fixed In:� |
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 |
In order to correctly process 1pcc calls to a PSTN destination in Alcatel IMS deployments, the following options must be configured on the originating DN:
make-call-rfc3725 to a value of 2.sip-ring-tone-mode to a value of 1.| Found In: 8.0.300.43 | Fixed In: 8.0.400.25 � |
For predictive calls to be routed correctly in IMS deployments, you must configure the SIP Server application with the option p-asserted-identity, using the same value for the asserted identity as used for the Routing Point DN.
(ER# 254989473)
| Found In: 8.0.300.42 | Fixed In: 8.0.400.25 |
SIP Server can sometimes undergo significant performance degradation related to the storage of postponed EventEstablished events in greeting scenarios.
(ER# 251763723)
| Found In: 8.0.300.34 | Fixed In: 8.0.300.37 |
Chat delivery through the T-Library connection (enabled by setting sip-signaling-chat to none) is intended for use with SIP endpoints only. In this case, the SIP endpoint should register with SIP Server. It will handle voice calls, with the chat interaction occurring by means of the T-Library connection. (ER# 248975305)
| Found In: 8.0.300.34 | Fixed In:� |
The Page mode for IM (Instant Messaging) is not supported. (ER# 247315792, 248975228)
| Found In: 8.0.300.34 | Fixed In:� |
SIP Server does not generate an EventRetrieved message in response to a TAlternateCall request in scenarios where GVP is configured as an MSML Media Server and cannot locate the file to be played as Music on Hold. (ER# 250926651)
| Found In: 8.0.300.34 | Fixed In: 8.0.400.25 |
SIP Server does not support REFER requests sent during the Early Dialog state. (ER# 249508823)
| Found In: 8.0.300.34 | Fixed In: 8.0.400.25 |
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:� |
If the SIP Server connection with an MCU fails in scenarios involving call recording, further call operations (hold, transfer, conference) are not possible. In this case, the call can be released only. (ER# 249358013)
| Found In: 8.0.300.34 | Fixed In: 8.0.400.25� |
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 |
If SIP Server receives a 1pcc re-INVITE with the Replaces header from one of the two parties on the call when both agent and caller greetings are in progress, SIP Server may drop the call.
(ER# 250790141)
| Found In: 8.0.300.34 | Fixed In: 8.0.400.31 � |
SIP Server may drop all conference members if, during an attempt to add a new participant to a conference, it receives an error response code 444 (Number of media inputs exceeded) from the Genesys Media Server. This situation can occur when the GVP Media Control Platform (MCP) option resource-confmaxsize is set to 32. (ER# 250802005)
| Found In: 8.0.300.34 | Fixed In: 8.0.400.25 |
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:� |
The Caller ID restriction feature provided through the dial-plan (dial-plan parameter clir=on) works with a new call, but does not fully work with a transfer or conference to a new party. When transferring or conferencing to a new party using the dial-plan, the existing parties on the call will use the clir setting that was provided when they joined the call, instead of the setting provided by the dial-plan-rule.(ER# 235360404, 248341724)
| Found In: 8.0.200.34 | Fixed In: 8.0.400.25 |
Use of the dial-plan parameter calltype can conflict with the DN-level option override-call-type when configured on a DN of type Routing Point. For calls to a Routing Point, Genesys recommends that you do not combine the use of these two options, otherwise the CallType attribute might change midway through the call. (ER# 234758271)
| Found In: 8.0.200.34 | Fixed In:�8.0.300.34 |
SIP Server does not request GVP to record the Call Progress Detection (CPD) phase of a predictive call if the CPD analysis is performed on the media gateway, even though the TMakePredictiveCall request contains the cpd-record key-value pair set to on in the Extensions attribute. (ER 238357601)
| Found In: 8.0.200.34 | Fixed In: 8.0.300.37 |
In certain scenarios, if an HA switchover takes place while an
INVITE transaction was in progress, SIP Server does not send the
CANCEL request to the same location as where the original
INVITE was sent. This issue can occur in the following
scenarios:
request-uri option for the DN is configured with a
different host than the host configured in the contact option
on the same DN.INVITE request
contains a Contact header with a URI that is different from the
INVITE destination.(ER�233704551)
| Found In:�8.0.200.34 | Fixed In: 8.0.400.25 |
The sip-ip-tos configuration option setting takes effect only after the SIP Server restart. (ER# 238135561)
| Found In: 8.0.100.21 | Fixed In: 8.0.300.34 |
SIP Server does not take into account the value of the
reuse-sdp-on-reinvite option during the TRetrieveCall
operation if both call participants are configured with
reuse-sdp-on-reinvite set to true. (ER# 231207441)
| Found In:�8.0.100.16 | Fixed In:�8.0.100.17 |
Changes to the sip-enable-moh option take effect as follows:
sip-enable-moh is updated dynamically at the DN-level,
changes do not take effect until the next new call.sip-enable-moh is configured at the DN-level, but the
value is empty, changes made to the application-level option do not take
effect until after the application is restarted.(ER# 228264731)
| Found In:�8.0.100.16 | Fixed In:� |
When sending a REFER message to the other party, SIP Server incorrectly sends a Max-Forwards value of 6, instead of the expected value as described in the RFC. (ER# 229815568)
| Found In: 8.0.100.12 | Fixed In: 8.0.400.25 � |
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 inadvertently drop the call if a
CompleteConference attempt fails because Stream Manager terminates
the dialog with a BYE request and the header Reason "No
matching codecs found". (ER# 229946401)
| Found In:�8.0.000.16 | Fixed In: 8.0.300.40� |
SIP Server delays execution of the CompleteConference request
in the following scenario:
INVITE from the first party is received in the primary
call.INVITE sequence is completed.INVITE sequence is completed.CompleteConference request is delayed until
no new requests remain for the primary call.| Found In:�8.0.000.16 | Fixed In: 8.0.000.17 |
SIP Server no longer supports chat functionality using Microsoft Live Communications Server (LCS). (ER# 228183840)
| Found In:�8.0.000.16 | Fixed In:� |
SIP Server is unable to dynamically update the DN-level options
sip-enable-moh and request-uri. Changes to either of
these options will not take effect until after the SIP Server is restarted.
(ER# 221245281)
| Found In:�8.0.000.12 | Fixed In: 8.0.400.25 |
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 |
SIP Server may issue unnecessary Standard log messages
(00-06080) during the startup initialization of the SIP Server.
(ER# 221891011)
| Found In:�8.0.000.12 | Fixed In:�8.0.000.16 |
SIP Server may inadvertently drop the main call when an agent initiates a
TSingleStepConference request and SIP Server receives a 486
Busy Here message from the media server. (ER# 219139759)
| Found In:�8.0.000.12 | Fixed In:�8.0.100.16 |
SIP Server does not support the Graceful Shutdown feature that was introduced in Genesys Administrator version 8.1.
| Found In:�8.0.000.12 | Fixed In:� |
SIP Server sometimes fails to correctly synchronize SUBSCRIPTION requests from Real Time Communication (RTC)-based user agents to the backup SIP Server instance. This can cause the subscription to stop working properly after the switchover until it renewed. (ER# 230813810)
| Found In: 7.6.000.77 | Fixed In:�8.0.200.35 |
While processing TMakePredctiveCall requests, SIP Server might
start recording too early in the call—at the
EventRouteRequest rather than at the moment when the call is
established on the agent phone. This issue can occur when the
record option is set to true on the
Trunk DN that represents the outbound gateway. (ER# 229724438)
| Found In:�7.6.000.76 | Fixed In:�8.0.300.34 |
When operating in a high-availability environment, after a switchover, SIP
Server may report a DN configured with the
use-register-for-service-state option set to true as
out of service. This issue occurs only if the DN was in the
in-service state before the backup SIP Server started and no
activity was reported on that DN. (ER# 217632696)
| Found In:�7.6.000.69 | Fixed In:�8.0.100.16 |
SIP Server cannot perform the alternate call operation to a call at a
Routing Point from a SIP endpoint for which the sip-cti-control
option must be set to hold. (ER# 214700427)
| Found In: 7.6.000.63 | Fixed In: 8.0.400.32 � |
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� |
A call party does not see "pushed" video when the AgentVideo
parameter in the Extensions attribute is set to
from-third-party. (ER# 171694535)
| Found In:�7.6.000.40 | Fixed In: 8.0.300.34� |
SIP Server may incorrectly update capacity information if active calls are present on a trunk. Genesys recommends that, for the changes to take effect, you restart SIP Server after you complete the capacity configuration. (ER# 172370691)
| Found In:�7.6.000.40 | Fixed In:�8.0.300.34 |
SIP Server cannot process a conference back to GVP (Genesys Voice Platform)
when the request-uri and From headers contain the
same DN numbers. (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 may not automatically release a monitored call when the following conditions are true:
MonitorScope is set to call and
MonitorMode is set to coach.To complete this call, either the second agent or the supervisor should release the call. (ER# 185899321)
| Found In:�7.6.000.40 | Fixed In: 8.0.400.25 |
SIP Server does not report an EventPartyDeleted message for a
DN associated with the remote supervisor in the following scenario:
(ER# 186379184)
| 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 does not send an INVITE message to the second
instance of the recorder service if the first instance of the recorder service
fails. A call will be established without the recorder service. (ER# 189554684)
| Found In:�7.6.000.40 | Fixed In:�8.0.100.16 |
SIP Server does not generate an EventEstablished message for
DN1 in the following scenario:
(ER# 190252826)
| Found In:�7.6.000.40 | Fixed In:�8.0.300.34 |
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 message with an empty body could disrupt a chat session when using the Instant Messenger. (ER# 114869267)
| Found In:�7.5.000.15 | Fixed In:�8.0.300.34 |
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:� |
The treatment PlayAnnouncementAndCollectDigits ends if digits
collection has completed because the MAX_DIGITS limit has been
reached or because the ABORT/TERM_DIGITS has been entered. This
scenario will cause an interruption of the announcement even if the
INTERRUPTABLE flag is set for this announcement. (ER# 20014599)
| Found In:�7.1.001.09 | Fixed In:� |
SIP Server mistakenly distributes a DNBackInService event if
the properties of the corresponding DN are changed in Configuration Manager.
(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.
SIP Server no longer supports the Windows 2000, 32-bit operating system.
| Discontinued As Of:�8.0.000.12 |
Information in this section is included for international customers.
There are no known internationalization issues for this product.
Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. The following documentation also contains information about this software. Please consult the Framework 8.0 SIP Server Deployment Guide first.
Framework 8.0 SIP Server Deployment Guide contains detailed reference information for the Genesys Framework 8.0 SIP Server, including configuration options and specific functionality.
Framework 8.0 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 8.0 SIP Server Integration Reference Manual contains reference information related to integrating SIP Server with SIP softswitches and gateways.
Framework 8.0 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 Technical Support for additional information.
Genesys Supported Operating Environment Reference Guide contains information about supported operating systems and databases.
Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD (produced monthly).
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. To determine the version of a document, check the version number that is located on the second page in PDFs or on the About This File topic in Help files.