Release Note

Contents

Introduction

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is no longer an affiliate of Alcatel-Lucent; any indication of such affiliation within Genesys products or packaging is no longer applicable.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.genesyslab.com for more details.

This release note applies to all 7.6 releases of SIP Server.

• Links in the Contents section enable you to access information regarding a specific release. Releases are listed by version number rather than in date order. For this reason, a recent release may be listed after earlier releases, if the version number is lower. Except when otherwise noted in the information for a specific release, each release includes all of the features and corrections that were introduced for the applicable operating system at earlier dates, regardless of the version numbering.
• The Release Number table lists the supported operating systems for each release at a high level. For more detailed information, see the Genesys Supported Operating Environment Reference Guide.
• The Known Issues and Recommendations section is a cumulative list. It includes information on when individual items were found and, if applicable, corrected.
• The Corrections and Modifications section for each release may list additional issues that were corrected without first being documented as Known Issues.
• The Discontinued Support section is cumulative. It lists functionality that is no longer supported and the release number of the version in which support was discontinued.
• You can find information about changes and corrections to functionality common to all T-Servers in the release note for T-Server Common Part release 7.6.x.

Use of Third-Party Software

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 Customer Care representative if you have any questions.

Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

In HA environments, memory allocations of a certain size (512 bytes) no longer cause unlimited memory consumption in a backup SIP Server. (SIP-5652)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

SIP Server no longer terminates while processing a response to INFO and INVITE requests if the INFO request is received before the dialog is established. SIP Server now starts processing the response to the INVITE request after receiving the final response to the INFO request. (ER# 303861494)

SIP Server now updates the contact parameter in the 200 OK response for the REGISTER message when a modified REGISTER request is received. Previously, SIP Server did not update the contact parameter while forming the response for the REGISTER refresh. Instead, SIP Server updated the contact parameter only after the REGISTER request expired. (ER# 309957404)

SIP Server no longer terminates while processing an invalid Instant Messaging scenario, which previously led to a loop of INVITE messages. SIP Server now correctly clears the erroneously created dialogs. (ER# 305304122)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

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# 298587120, 299700997)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

When a call is rejected because of the Ring-Through Rules functionality (reject-call-incall=true), SIP Server now sends a 486 Busy Here error response to indicate that the DN is busy with another call. Previously in this scenario, SIP Server sent a 481 Call Leg/Transaction Does Not Exist error response. (ER# 292640278)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now successfully processes INFO messages that are sent while processing INVITE transactions for call setup. Previously in this scenario, the 200 OK response to the INVITE was ignored. (ER# 291686088)

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Supported Operating Systems

New in This Release

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 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.

• SIP Server now supports setting the option sip-enable-moh to na on Voice Treatment Port DNs. This setting delays the 200 OK message that SIP Server sends in response to a 1pcc hold operation. Instead, SIP Server sends a hold request to the peer party and only after receiving a response from the peer party will it send the 200 OK message to the original Hold request.

  SIP Server does not send the T-Library messages EventHeld or EventRetrieved for 1pcc hold/retrieve operations for Voice Treatment Ports with sip-enable-moh set to na.

  (ER# 289537514)

Corrections and Modifications

This release includes no corrections or modifications.

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Supported Operating Systems

New in This Release

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 release does not contain any new features or functionality.

Corrections and Modifications

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# 271740106, 269458106)

SIP Server now correctly processes a single-step conference scenario in which the conference initiator goes to the out-of-service state during the establishing phase of the conference. Previously in this scenario, SIP Server sometimes became unstable. (ER# 272670696)

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Supported Operating Systems

New in This Release

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 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.

• When working with multiple devices, the geo-location mechanism now works in call transfer scenarios using the REFER method as follows:

  1. An inbound call from an external caller matches a Trunk DN, for which the geo-location option is set.
  2. The call is routed to an internal DN.
  3. By using the REFER method, the internal DN refers the call to an external DN—that is, SIP Server selects the outbound trunk with the same geo-location option value as the incoming trunk's geo-location value.

  (ER# 267159606)

• SIP Server now supports SIP Session refresh requests from a media server to SIP Server in the following scenario:
  
  When a media server sends a re-INVITE request with an SDP offer within a dialog, SIP Server responds with a 200 OK message including an SDP answer. Previously, SIP Server incorrectly sent a 500 Server Internal Error message as a response.

  (ER# 266943686)

Corrections and Modifications

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 redirected the call to a no-answer-overflow DN when the no-answer timeout expired. (ER# 268116465)

SIP Server now correctly distributes an EventTreatmentNotApplied message for a failed TApplyTreatment request in the following scenario:

1. A call arrives at a Routing Point.
2. URS successfully applies a treatment to the call.
3. While the first treatment is in progress, URS attempts to apply another treatment to the call.
4. A media server rejects the INVITE request to apply the treatment.

SIP Server now correctly reports a CallState attribute set to 22 (Redirected) in the EventRouteUsed message, which it generates when router-timeout expires and the default routing destination specified in the default-dn option is out of service. (ER# 268714096)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now sends an EventRouteUsed message and drops the call if router-timeout expires and the default routing destination specified in the default-dn option is out of service. Previously, SIP Server did not process such a scenario correctly, which resulted in a stuck call. (ER# 266747874)

Note: Currently, there is a Known Issue regarding a missing AttributeCallState, which should be included in the EventRouteUsed. See ER# 268714096.

SIP Server no longer becomes unstable when attempting to play a busy tone but a treatment service DN (DN of type Voice over IP Service) has not been configured. SIP Server will now drop the call in this scenario. (ER# 267246187)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly handles the following situation: there is no response to a re-INVITE request when SIP Server is promoted to Backup Mode. Previously, SIP Server might become unstable in this situation. (ER# 265828729)

Genesys has implemented a change in SIP Server behavior called Route loop prevention. Route loop prevention calls for SIP Server to drop a call if Universal Routing Server (URS) attempts to route that call more than 100 times in connection with a single interaction. This modification prevents URS from endlessly trying to route a call when SIP Server returns EventError after an unsuccessful route attempt, for example, EventError �Object not known,� but excluding EventError for request validation, for example, �Invalid DN.� The related counter is reset if the route attempt is successful, even when routing to another Route Point. When SIP Server drops a call according to this behavior, it raises the following alarm:

Level: Standard
Name: GCTI_SIP_CALL_TERMINATED
id: 52024
Text: Call [call] was unexpectedly terminated by SIPServer with reason [reason]
Attributes:
[call] � the conn-id of the terminated call.
[reason] � reason the call was terminated. (will be �too many routing attempts� for this feature).
Description: SIPServer has been forced to terminate an active call for the reason stated in the message.

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly releases a consultation call and sends an EventAbandoned message in the scenario where, after some unsuccessful attempts to make a consultation call to a Routing Point, an agent releases the consultation call and reconnects to the main call. Previously in this scenario, SIP Server did not send EventAbandoned to URS, which prevented the consultation call from being released completely. (ER# 265492788)

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# 265672984)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server no longer becomes unstable while attempting to process a TMakeCall scenario where the call is made from a DN that was deleted from the configuration environment during the call. (ER# 263392363)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly handles monitoring sessions in the following scenario:

• A supervisor sends a RequestMonitorNextCall for a call already established between the caller and an agent.
• The supervisor device is invited to join the monitoring session, but it responds with a 180 Ringing message.
• The option sip-ring-tone-mode in the SIP Server application is set to 1.

SIP Server no longer becomes unstable while handling a race-condition scenario in which TInitiateTransfer and TSingleStepTransfer requests are sent simultaneously from the same DN. (ER# 262839861)

While placing an endpoint on hold, SIP Server no longer creates an incorrect hold SDP when processing a multipart body of the SIP message. Previously in this scenario, SIP Server sometimes created an incorrect hold SDP, which resulted in the hold re-INVITE being rejected by the endpoint. (ER# 229391911)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server no longer adds an additional P-Charging-Vector header in outgoing SIP messages. Previously, SIP Server would include duplicated P-Charging-Vector headers, which affected the call flow. (ER# 257351536)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server no longer encounters the stuck call situation that is described in the following scenario:

1. A RequestInitiateTransfer or RequestInitiateConference is issued.
2. The main call is immediately released with a RequestReleaseCall, but before the EventDialing was reported on the consultation call.
3. Call data for the consultation call was retained in the SIP Server memory.

SIP Server now uses the round-robin method to choose a destination from a pool of available trunks, in cases where the routing (transfer) of an outbound call takes SIP Server out of the signaling path. Previously in this scenario, SIP Server selected the first available trunk. (ER# 256698144)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

In multi-site scenarios, where SIP Server uses INVITE messages with the Replaces header, SIP Server now correctly generates EventUserEvent messages with a complete list of call participants. Previously, SIP Server did not generate such messages, which caused T-Library clients that use those messages to display an incomplete list of call participants. (ER# 250322240)

When operating with the BroadSoft BroadWorks switch, SIP Server now places an agent in the Not Ready state whenever SIP Server is notified by the switch that some call activity has begun on the phone's private line, even if the agent has another active call on the business line. If the call on the private line is still active when the business call ends, the agent will remain in the Not Ready state until the end of the private call. Previously, with an active call on the business line, SIP Server did not take into consideration any call activity on the private line.

Note: If a private call on a private line ends during the After Call Work timeout, the agent becomes ready and the After Call Work period ends before the timeout expires. If a private call on a business line (it does not arrive via SIP Server) starts during an active SIP Server call, the agent will be set to the Not Ready state with the Unknown mode (avoiding After Call Work) immediately after the SIP Server call is released. The agent will be set to the Ready state when the private call ends.

(ER# 253950709)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

The DN-level valid values of the sip-enable-moh configuration option have been modified as follows:

sip-enable-moh
Default Value: No default value (Application-level setting applies)
Valid Values: true, false, na
Changes Take Effect: At the next call

If this option is set to true in the device configuration, it enables the music-on-hold treatment for any party that is engaged with this device in the call. If this option is set to false, it disables the music-on-hold treatment for any party that is engaged with this device in the call, even if the device sends an INVITE request containing a hold SDP.

The na value can be used only on Trunk DNs. If it is used for the trunks connecting SIP Servers in a multi-site environment, and if a call goes through multiple SIP Servers, then only the origination SIP Server, which received a hold SDP from the endpoint, will process the hold SDP and involve the media server to play Music On Hold. All other SIP Servers in the call path will propagate the hold SDP to the destination without involving the media server.

Note: The DN-level option takes precedence over the Application-level option.

(ER# 250919774)

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Supported Operating Systems

New in This Release

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 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# 250332812)

• 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# 250332638)

Corrections and Modifications

This release includes the following corrections or modifications:

SIP Server now properly extracts part of the XML message related to a particular entity from the NOTIFY message that arrives from the BroadSoft BroadWorks switch. Previously, SIP Server sometimes extracted an incorrect part of the XML message, leading to inaccurate reporting of agent state updates. (ER# 250598608)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

SIP Server now releases the party from a call if this party, while responding with a SIP error message to the INVITE request sent by SIP Server, continues sending other T-Library requests. Previously in this scenario, SIP Server became unstable. (ER# 248476328)

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:

• On application startup, when the enabled/disabled DN is read.
• While the application is running, when a DN is enabled/disabled in Configuration Manager, or when a new enabled/disabled DN is first created.

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 did not distribute an EventCallPartyAdded message when a conference was completed to the same external destination. (ER# 248848158)

SIP Server no longer becomes unstable while providing a ringback to the device on the consultation call if the device responds with the SDP that does not contain the codecs specified in the audio-codecs configuration option. This issue occurred if the device was configured with the dual-dialog-enabled configuration option set to false, and SIP Server was configured with the sip-enable-sdp-codec-filter option set to true. (ER# 241900677)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

SIP Server now accepts the 302 Moved Temporarily message received from GVP Resource Manager (RM) even if a caller cancels the call before this message arrives. SIP Server also sends a NOTIFY message to RM containing the terminated state to free IPCS ports in RM. The NOTIFY message body will now contain only the entity value and will no longer contain the call-id value. (ER# 245563971)

SIP Server now terminates a SIP dialog upon receiving a 481 Call/Transaction Does Not Exist response to the re-INVITE message. Previously in this scenario, SIP Server did not terminate the SIP dialog, which resulted in a stuck call. (ER# 244456005)

SIP Server now terminates a SIP dialog if it receives only the 100 Trying response to the re-INVITE message. To support this, SIP Server starts a timer when a 100 Trying response arrives. This timer is set to the value of option sip-invite-timeout or it is set to 32 seconds if that option is empty or is not configured. This timer is reset by any event related to this dialog. If the timer expires, SIP Server terminates such dialog. Previously in this scenario, SIP Server did not terminate the dialog, which resulted in a stuck call. (ER# 245966662)

When running in a backup mode, SIP Server no longer grows in memory while processing calls with personal greetings. Previously, the backup SIP Server memory utilization increased in this scenario. (ER# 246975719)

SIP Server now always provides a value in the Content-Type header of a SIP request. Previously, starting with version 7.6.001.06, SIP Server might have sent a SIP request with an empty Content-Type header and some SIP clients might have rejected such a request. (ER# 247074744)

When running in a backup mode, SIP Server no longer grows in memory in a scenario in which an inbound call, through SIP Server 1, is connected to an agent on SIP Server 2, and the agent submits a TInitiateTransfer request. The backup SIP Server memory utilization increased only if the sip-enable-moh configuration option was set to true. (ER# 244730111)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or modifications:

SIP Server no longer drops calls if the caller sends a re-INVITE before the destination answers the call. (ER# 288242245)

SIP Server now correctly processes the following scenario:

1. A call from an external caller matches a Trunk DN, for which the geo-location option is set. The call arrives at a Routing Point.
2. URS routes the call to an external number.
3. SIP Server selects the outbound trunk with the same geo-location option value as the incoming trunk's geo-location value.

SIP Server now correctly processes the following scenario:

1. SIP Sever propagates an INVITE message from an originating party to a destination.
2. The destination responds with a 18X message containing SDP.
3. SIP Server sends a PRACK message to the destination and propagates the 18X message containing SDP to the originating party.
4. Before the originating party sends 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.

When operating in a high-availability environment, after a switchover, SIP Server now places a DN configured with the use-register-for-service-state option set to true in the out-of-service state if SIP Server does not receive a new REGISTER request from that DN within a registration timeout. Previously, SIP Server did not place such a DN in the out-of-service state. (ER# 243946908)

When two SIP Servers negotiate a session refresh interval within one INVITE transaction, they now choose different values for session timers, as follows: the session timer at the server side of the INVITE transaction will be 6 seconds longer than the session timer at the client side of the transaction. The refresh re-INVITE request from the client will discard the refresh re-INVITE from the server. As an additional precaution against the synchronization of client's refresh, Genesys recommends configuring different values of the session-refresh-interval configuration option for each SIP Server. (ER# 244292760)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections or 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# 245502540)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release contains the following corrections or modifications:

While processing a call to an external Routing Point by means of ISCC, SIP Server now correctly sends the call to the external Routing Point. Previously, SIP Server incorrectly sent the call to the Routing Point, which was specified by the default-route-point configuration option. (ER# 238344100)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release contains the following corrections or modifications:

A new DN-level configuration option, sip-oos-enabled, has been added in this release.

sip-oos-enabled
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately

If this option is set to true, SIP Server will place the corresponding device in the out-of-service state for not responding to SIP requests. If this option is set to false, SIP Server will not place the device in the out-of-service state for not responding to SIP requests. (ER# 244202631)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release contains the following corrections or modifications:

SIP Server now correctly processes a 491 Request Pending response to the INVITE message by starting a timer and resending the same INVITE request when the timer expires. Previously, SIP Server did not handle the 491 Request Pending response correctly. (ER# 242606037)

When GVP Resource Manager (RM) sends back a 302 Moved Temporarily message with the Contact of the IPCS along with the GVP-Resource-ID parameter, SIP Server now attaches it as UserData to the call, and also sends the GVP-Resource-ID in the Request-URI of the INVITE message after processing the TRouteCall request. In addition, SIP Server will send a NOTIFY message with the GVP-Resource-ID parameter to RM to notify whether a port is available or occupied. RM uses the GVP-Resource-ID parameter for its internal port management. (ER# 241815513)

In a scenario where a client is subscribed for call supervision on multiple registered DNs, and then one of the DNs is unregistered because the cancel-monitor-on-disconnect option is set to true, SIP Server now cancels call monitoring on that particular DN. Previously, SIP Server would cancel call monitoring on all of the client's subscribed DNs. (ER# 242882315, 244870004)

SIP Server now correctly processes TMakeCall requests on behalf of a DN in the following scenario:

1. DN1 is involved in a consultation call.
2. The consultation call is released, and the main call (involving DN1) is retrieved.
3. DN1 sends a TMakeCall request.

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Supported Operating Systems

New in This Release

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 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 Extensions attribute with the CPNDigits key is now supported in TRouteCall requests.
  
  When the CPNDigits key is set to a specific value in the Extensions attribute, the value overrides the username provided in the URI in the From header of the INVITE message. (ER# 232492185)

Corrections and Modifications

This release contains the following corrections or modifications:

When propagating an INFO message from DMX to a treatment service, SIP Server now correctly treats this message as a DTMF digit. Previously, SIP Server ignored it. This issue occurred while SIP Server was processing the PlayAnnouncementAndDigits voice treatment. (ER# 239305799)

SIP Server no longer generates an EventAgentNotReady message on receiving a NOTIFY message that indicates a Confirmed state in cases where there is a call routed to an agent. (ER# 239560651)

SIP Server now correctly applies a Busy treatment if a call reaches a device that responds with the 486 Busy Here message. Previously in this scenario, SIP Server applied a Fast Busy treatment. (ER# 240631909)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release contains the following corrections or modifications:

SIP Server now correctly handles call scenarios where an HA switchover occurs while the call is in the ringing state, and the caller subsequently releases the call. Previously, SIP Server was unable to send the CANCEL request for the released call. (ER# 244626981)

In consultation call scenarios where the main call is on hold, the consultation call is established, and the device for the agent who initiated the call disconnects from the network, SIP Server now detects that this device is unresponsive (when the session-refresh-interval expires), allowing SIP Server to correctly terminate all dialogs for both the main and consultation calls. Previously, SIP Server did not terminate the main call dialogs and the caller remained connected to the device that provided the Music-On-Hold service. This issue occurred if the DN was configured with the dual-dialog-enabled configuration option set to false. (ER# 235707869)

In a monitored call, if an agent phone becomes unresponsive (is shut down or unplugged from the network), SIP Server now disconnects all parties on the call at the next session refresh, when SIP Server places the agent DN in out-of-service state. Previously, the calling party remained on the call. (ER# 239515956)

SIP Server now properly releases a DN that is involved in silent voice monitoring if this DN becomes unresponsive because of a network failure. Previously, SIP Server sometimes became unstable while releasing such a DN. (ER# 239515989)

SIP Server now abandons the call if the timer expires after a TRouteCall request because the call's inbound leg did not send a final response to the re-INVITE request. Previously, SIP Server kept such a call active and rejected all consecutive TRouteCall requests with an EventError message. This led to incorrect multiple allocation of call center agents for that call. This issue occurred if the routing destination DN was configured with the reuse-sdp-on-reinvite configuration option set to true. (ER# 236814473)

When receiving a SIP OPTIONS message with the Max-Forwards header containing a value of 1, SIP Server now correctly responds with the 200 OK message. Previously, SIP Server responded with the 483 Too Many Hops message. (ER# 238515097)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release contains the following corrections or modifications:

SIP Server now properly retrieves the main call when processing a TReconnectCall request, even if the other party of the consultation call disconnects from the call at the same time. Previously in this scenario, SIP Server did not retrieve the main call. (ER# 235257379)

SIP Server now correctly processes the following scenario:

1. A call is established between a customer and an agent located behind the softswitch.
2. A DN of type Voice over IP Service with service-type set to softswitch that is used to access the agent's extension is taken out of service.
3. The agent submits the TInitiateConference request.

SIP Server now correctly applies AfterCallWork (ACW) functionality to outbound calls. In previous versions of 7.6 SIP Server, this functionality was broken. (ER# 228979595)

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Supported Operating Systems

New in This Release

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 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 route inbound calls initiated by incoming INVITE requests to a specified DN. To support this feature, a new Application-level configuration option, default-route-point, has been added.
  
  default-route-point
  Default value: An empty string
  Valid value: Any valid DN
  Changes Take Effect: For the next call
  
  Specifies the DN to which an inbound call is routed when its destination number is external. For this call, SIP Server will report the call's original destination number taken from the Request-URI of the INVITE message in the DNIS attribute of the EventRouteRequest message. If the default-route-point option is not configured (or does not have any value), then inbound calls are handled in accordance with the regular procedure.
  
  Note: This functionality is applicable only to inbound calls initiated by incoming INVITE requests. This functionality is not applicable to ISCC calls and to calls initiated by T-Library requests, such as TMakeCall, TInitiateTransfer, and so on.
  
  (ER# 236303941)

Corrections and Modifications

This release does not contain any corrections or modifications.

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now properly handles a T-Library client request TQueryAddress with the AddressInfoType parameter set to CallsQuery, which queries DNs with AddressType set to Queue or RouteDN. Previously, SIP Server sometimes became unstable if this request arrived when 100 or more active calls were on that DN. (ER# 235285157)

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Supported Operating Systems

New in This Release

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 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.

• SIP Server now supports the Network Attended Transfer/Conference NAT/C feature, which allows agents working in multi-site contact centers to consult with each other before making call transfers or conferences. For more information about this feature, see the "Network Attended Transfer/Conference Feature" section in the "Multi-Site Support" chapter of the Framework 7.6 SIP Server Deployment Guide.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now rejects a TCompleteConference or TCompleteTransfer request with an error if a party from a consultation call that is to be merged into the main call has the same number as one of the parties from the main call. Previously, SIP Server accepted such requests, which caused SIP Server memory utilization to grow, which in turn caused SIP Server to become unstable. (ER# 233687981, 240141754, 226791350)

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)

When operating in hot standby mode after a switchover, the new primary SIP Server now correctly distributes a Call Monitoring event. Previously in this scenario, SIP Server did not distribute a Call Monitoring event until a new RequestStartCallMonitoring message arrived. (ER# 233740405)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

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 when the HTTP stream is disconnected. SIP Server selects the next available media server (for example, an instance of Stream Manager) in a round-robin fashion. If a server fails to start at a particular Stream Manager instance, SIP Server tries the next instance, but if the server 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 media server had previously failed. (ER# 233377191)

SIP Server now always propagates the AttributeReferenceID and AttributeReason parameters obtained from the TSetAgentNotReady request into the corresponding EventAgentNotReady message. Previously, a TMakeCall request that was invoked toward a particular DN sometimes prevented distribution of these attributes in EventAgentNotReady for that DN. (ER# 233396771)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now properly processes the following scenario:

1. A call is established, and SIP Server receives a re-INVITE request with a hold SDP from one of the SIP call legs.
2. SIP Server responds with a 200 OK message and receives an ACK message in return.
3. SIP Server initiates a hold INVITE transaction with a second call leg.
4. SIP Server receives a new re-INVITE request from the first call leg while the re-INVITE transaction is still in progress. SIP Server responds to the re-INVITE with a 200 OK (hold SDP).
5. When both re-INVITE transactions (Step 3 and Step 4) are acknowledged, SIP Server initiates a new re-INVITE message exchange between call legs, restoring the audio path between them.

Previously, SIP Server did not complete Step 5 in this scenario. (ER#  231902049, 232051753)

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# 229962765)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly applies the make-call-alert-info configuration option setting and includes the Alert-Info header inside the initial INVITE message that is sent to the origination party in response to a TMakeCall request. Previously, SIP Server did not include the Alert-Info header in the INVITE message. This issue occurred in SIP Server version 7.6.000.40 and later. (ER# 229729022)

SIP Server now sends a 200 (OK) message in response to a gateway's BYE message when routing using the REFER method is finished. Previously, SIP Server did not send any response to the gateway's BYE message. This issue occurred if a Trunk DN representing the gateway was configured with the oosp-transfer-enabled configuration option set to true. (ER# 230041160)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now properly completes a call transfer in the following scenario:

1. An inbound call at Site A is routed to Agent 1.
2. Agent 1 at Site A initiates a consultation call to Agent 2 at Site B.
3. Agent 1 at Site A completes the call transfer.
4. Agent 2 at Site B initiates a consultation call to Agent 3, back at Site A.
5. Agent 2 at Site B completes the call transfer.
6. The inbound call is connected to Agent 3 at Site A.

SIP Server no longer becomes unstable when a call is released at a DN that had been deleted in Configuration Manager while the call was active, sending the BYE message to the deleted DN. This issue occurred if a client was registered for Voice RTP Monitoring at that DN. (ER# 230027242)

SIP Server now supports recovery-timeout functionality for the following types of DNs: Extension, ACD Position, and Voice Treatment Port. When the DN is placed out of service for not responding to the INVITE request, SIP Server is now able to place these DNs back in service after a configured recovery-timeout period expires. Previously, these DNs remained out of service even after the recovery-timeout expired. (ER# 230791242)

SIP Server now releases an inbound call at a Routing Point if the inbound gateway does not send an ACK message in response to the 200 (OK) within 32 seconds after the routing is complete. Previously, SIP Server attempted to reroute the call and, eventually, became unstable. This issue occurred if SIP Server was configured with the event-ringing-on-100trying configuration option set to true. (ER# 230220909)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

When performing a trunk optimization by a REFER method in a call transfer scenario, SIP Server now correctly selects a trunk to another SIP Server based on the IP address and port taken from the trunk configuration. Previously, the trunk was chosen incorrectly among other trunks if they were configured on the switch without prefixes, and thus, the INVITE request was sent to the wrong destination. As a result, SIP Server could not complete the transfer transaction. (ER# 229403131)

SIP Server now properly handles race conditions in the following scenario:

1. A consultation call is initiated and answered.
2. The first party receives a SIP re-INVITE request for the main call.
3. SIP Server receives a TCompleteConference request from an agent desktop before the re-INVITE sequence is completed.
4. The re-INVITE sequence is completed.

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server no longer removes the m= section from the OK SIP message when an inbound call, which is routed to an agent, is answered. This issue only occurred if SIP Server was configured with the sip-enable-sdp-codec-filter configuration option set to true, and the incoming INVITE request specified a Content-Type header as multipart/mixed but the message contained only a single part. Previously, SIP Server sometimes removed m= sections from the SDP and sent the OK message with the corrupted SDP to the origination party. (ER# 229075418)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now properly routes an inbound call, even if the message body in the incoming INVITE request specifies a Content-Type header as multipart/mixed, but the message contains only a single part, which is application/sdp. Previously, SIP Server rejected the TRouteCall request with the EventError message. This issue only occurred if SIP Server was configured with the sip-enable-sdp-codec-filter configuration option set to true. (ER# 225589039)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

When operating in warm standby mode, SIP Server now correctly closes the port on which it listens to incoming SIP requests when switching to backup mode, and opens this port when switching to primary mode. Previously after a switchover, SIP Server was unavailable for up to 20 seconds. (ER# 219315971)

When processing a TMakeCall request, SIP Server now correctly reports the CallState attribute in an EventDestinationBusy message. If a destination device replies with the 404 NotFound message, the CallState attribute is set to 11. If the destination device replies with the 486 Busy Here message, the CallState attribute is set to 6. Previously in both cases, SIP Server generated EventDestinationBusy containing the CallState attribute set to 6. (ER# 219714837)

SIP Server now restarts a voice treatment using an available voice treatment service if the media server, which processes this treatment, is placed in the out-of-service state, after SIP Server does not receive any response from this media server to the OPTIONS request. Previously in this scenario, SIP Server did not restart the voice treatment. (ER# 222237360) See Known Issues and Recommendations: ER# 222237360, 224997668.

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly processes the scenario where, during a call recording session that is initiated based on the TRouteCall request containing the record key set to source in the Extensions attribute, the call is transferred by an agent whose device is configured with the record configuration option set to true. Previously in this scenario, SIP Server terminated the recording session after the call transfer was completed. (ER# 221549652)

SIP Server no longer allows a supervisor to participate in more than one monitoring session at a time. (ER# 218570309)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now distributes an EventReleased message in which the CallState attribute is set to 7 (CallStateNoAnswer) when it releases an unanswered predictive call after the timeout specified in a TMakePredictiveCall request expires. Previously in this scenario, SIP Server distributed EventReleased in which the CallState attribute was set to 0 (CallStateOk). (ER# 216974315)

SIP Server now correctly processes the DIGIT_TIMEOUT parameter in TApplyTreatment requests containing the PlayAnnouncementAndDigits treatment type. (ER# 219128219)

When operating in hot standby mode, a backup SIP Server no longer sends polling OPTIONS requests to gateways or media servers that are configured with the active out-of-service detection feature enabled (the oos-check configuration option is set to a non-zero value). Previously, the backup SIP Server attempted to send OPTIONS requests to those devices, and it incorrectly marked those devices as out of service if no responses were received.

Now, after a switchover or startup, the primary SIP Server starts to send polling OPTIONS requests after a time interval equal to a value of the oos-check option expires. This time interval is necessary to account for possible network switching delays. Previously, the primary SIP Server sent polling OPTIONS requests immediately after a switchover or startup. (ER# 219334636)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server has been rebuilt to correct a minor build issue.

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Supported Operating Systems

New in This Release

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 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:

• Starting with this release, SIP Server can now request an offer from a calling device in a consultation call and send it to a destination DN, if the reuse-sdp-on-reinvite configuration option is set to true in the destination DN configuration. This feature is applicable to single-dialog DNs.
  
  The primary call could be disconnected if the signaling timeouts expire during the SIP offer-answer exchange. To prevent this, set the sip-invite-timeout configuration option to a value of less than 32 seconds. The recommended value range is 20–25 seconds. If the signaling timeouts expire, SIP Server will apply a busy tone to complete the offer-answer exchange.
  
  (ER# 216981264, 218279166)

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly requests an offer from a calling device in a scenario where a treatment is applied to a call, and the call is then routed to a destination DN that has the reuse-sdp-on-reinvite option set to true in its configuration. (ER# 219224131)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now initiates a new SIP SUBSCRIBE dialog if any of the following conditions occur:

• The refresh SUBSCRIBE request within the current SIP dialog is rejected or is not responded to.



• The BroadSoft switch sends a NOTIFY request containing the value terminated in the Subscription-State header.

A new Application-level configuration option has been introduced:

sip-retry-timeout
Default Value: 30
Valid Values: 1–3600
Changes Takes Effect: Immediately

This option specifies the time interval, in seconds, after which SIP Server initiates a new subscription if the previous SUBSCRIBE dialog is terminated.

(ER# 217704685)

SIP Server no longer disconnects the remote supervisor from a monitoring session when a monitored agent answers an incoming call. Previously, after SIP Server mistakenly disconnected the remote supervisor, any subsequent attempts to initialize a remote monitoring session also failed. (ER# 217563491, 217563505)

SIP Server no longer becomes unstable while processing a series of TSingleStepTransfer requests being sent by an agent desktop to an unknown destination. (ER# 214433878)

SIP Server now correctly processes a TCompleteConference request if no value has been specified for the contact option in the MCU (Multipoint Conference Unit) object configuration. Previously in this scenario, SIP Server produced a memory access violation. (ER# 180093529)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

If a c= line is present at both session and media levels in the SDP, SIP Server now correctly updates all c= lines in the hold SDP with the 0.0.0.0 IP address. Previously, SIP Server generated a hold SDP with the 0.0.0.0 IP address only at the session-level line, which contradicted the RFC 2543. (ER# 216170461)

In deployments where a gateway and agent SIP endpoints are configured to not provide provisional responses (messages starting with 18x) to initial INVITE messages, SIP Server now correctly handles the following scenario:

1. An outbound call is initiated based on a TMakePredictiveCall request.
2. A gateway responds with a 200 OK message.
3. While at a Routing Point, a treatment is applied to the call.
4. SIP Server routes the call to an available agent endpoint.
5. The agent endpoint responds with a 200 OK message.

If a supervisor's DN does not respond to INVITE requests during a call monitoring session initiation, while an established call is being recorded, SIP Server now places this DN in the out-of-service state when the session timeout expires, and cancels the call monitoring session. Previously in this scenario, after the session timeout expired, SIP Server did not place such a DN in the out-of-service state and sometimes produced a memory access violation. (ER# 155205489)

When disconnecting a call, SIP Server now sends a CANCEL request to the same address the original INVITE request is sent, even if a provisional response contains both the Contact header and the Record-Route header with the URI different from the URI containing in the original INVITE request. Previously, SIP Server mistakenly sent the CANCEL request to the address taken from the Record-Route header of the provisional response. (ER# 218578315)

SIP Server now correctly restores an audio path between the two remaining call participants when a conference has been completed or if the conference failed. Previously in some scenarios involving SIP endpoints with a different order of codec preference, a SIP offer-answer exchange was not performed correctly, which resulted in a broken audio path. (ER# 196118498)

For an inbound call being recorded, SIP Server now sends a re-INVITE request to an external call party when the session-refresh-interval timer expires. Previously in this scenario, SIP Server could become unstable after the external party terminated without sending the appropriate SIP messages. (ER# 181982800, 183499931)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly processes a SIP REGISTER request containing the Expires header set to 0 (zero) if this request is the first request of the SIP dialog. Previously, SIP Server ignored this request and did not make appropriate changes to DN and agent states. (ER# 142438546,  212726614)

When an agent is set to a Ready state by the endpoint's SIP PUBLISH request, SIP Server now accepts a TAgentNotReady request and generates an EventAgentNotReady message. Previously in this scenario, SIP Server ignored TAgentNotReady requests. (ER# 210408189)

When operating in a high-availability environment in hot standby mode and when endpoints are configured for Agent Presence, agent states are now correctly synchronized between primary and backup SIP Servers based on SIP PUBLISH messages containing Agent Presence information. (ER# 213443449)

SIP Server now generates an EventAgentNotReady message with the AgentWorkMode attribute set to Unknown when a SIP PUBLISH message, which corresponds to an endpoint's Busy status, arrives while an agent is still in the AfterCallWork state. Previously, SIP Server mistakenly generated EventAgentNotReady with the AgentWorkMode attribute set to AfterCallWork. (ER# 213524801)

When an agent changes the Availability status on his or her endpoint from Offline to Busy or Away, SIP Server now generates a sequence of EventAgentLogin and EventAgentNotReady messages. Previously, SIP Server ignored such agent endpoint status updates. (ER# 213731611)

When operating in a high-availability environment, a backup SIP Server no longer places a DN configured with the use-register-for-service-state option into the out-of-service state, based on Configuration Server updates. (ER# 213740951)

SIP Server now processes a SIP PUBLISH request and generates an EventAgentNotReady message when an agent changes the Availability status on his or her endpoint to Busy or Away while a call is in the established state on that endpoint. Previously in this scenario, SIP Server did not generate any event messages. (ER# 214095624)

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Supported Operating Systems

New in This Release

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 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:

• A new configuration option, sip-ring-tone-mode, is added to SIP Server. It works with the ring-tone-on-make-call option set to true in a scenario with a consultation call in a single dialog mode. The sip-ring-tone-mode option can be configured at the Application or Switch/DN level. The option setting at the Switch/DN level takes precedence over the Application level setting.
  
  sip-ring-tone-mode
  Default Value: 0
  Valid Values: 0, 1
  Changes Take Effect: Immediately
  
  With the option set to 1, when a called device requests an offer and the returned response cannot be used as the offer to a calling device, SIP Server connects Stream Manager to a call to play an audio ring tone.
  
  (ER# 212828575)



• A new Application-level option, set-notready-on-busy, is added to SIP Server.
  
  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 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)



• A new Application-level option, map-sip-errors, is added to SIP Server.
  
  map-sip-errors
  Default Value: true
  Valid Values: true, false
  Changes Take Effect: Immediately
  
  This option specifies whether a T-Library or SIP error code is reported in the ErrorCode attribute of an EventError message in response to a TRouteCall request. When set to false, the SIP status code is reported instead of the T-Library error code. For example, if routing is made to a busy destination, SIP Server will report the ErrorCode attribute of the EventError message as 486.



• SIP Server now supports two new DN-level options for DNs of type Trunk.
  
  display-name
  Default Value: No default value
  Valid Values: A string
  Changes Take Effect: Immediately
  
  This option specifies the string that will be sent as a display name in the From header of the INVITE request. This option applies only if the use-display-name configuration option is set to true.
  
  use-display-name
  Default Value: false
  Valid Values: true, false
  Changes Take Effect: Immediately
  
  This option activates the display-name configuration option.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly processes a scenario in which an inbound call that is parked at a Routing Point is dropped during a music treatment. Previously, SIP Server sometimes produced a memory access violation. (ER# 209034992)

When SIP Server receives DTMF tones for playing interruptible music treatment, it terminates the treatment once the Max Digits setting is reached. Previously, SIP Server sent a BYE message to the treatment device and, before the BYE transaction was completed, SIP Server mistakenly initiated a new SIP transaction with the treatment device which resulted in a 481 Call/Transaction Does Not Exist message. (ER# 175152192)

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Supported Operating Systems

New in This Release

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 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.

Corrections and Modifications

This release includes the following corrections and modifications:

SIP Server now correctly plays music for a call on hold when a THoldCall request is submitted by an agent endpoint. This issue occurred when SIP Server communicated with the agent endpoint via the ACME server. (ER# 212993755)

SIP Server now supports the REFER method for the first-party call control (1pcc) single-step transfer operation on the Snom 320 phones. (ER# 192831332)

SIP Server now responds to INFO messages received from Stream Manager (a recording device) with a 200 OK response. Previously, SIP Server did not respond to these messages. Re-transmission of INFO messages sometimes resulted in decreased SIP Server and Stream Manager performance. (ER# 136377863)

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Supported Operating Systems

New in This Release

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 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.

• SIP Server now supports the routing of an inbound 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. This feature applies to the following scenarios:
  
  
  • An inbound call is routed from a Routing Point to an external destination.
  • An agent transfers an inbound call by using the single-step transfer to a Routing Point, and then the call is routed to an external destination.
  • An agent transfers an inbound call by using the blind transfer to a Routing Point, and then the call is routed to an external destination.

  Note: This feature is not applicable for scenarios (the second and the third, above) where a conference (supervision or emergency recording) is involved.

  To support this feature, configure the following options for a DN of type Trunk:
  
  oosp-transfer-enabled=true
refer-enabled=true

  (ER# 197181309)

• SIP Server now distributes two new log messages and EventDNOutOfService and EventDNBackInService messages when a device of type Voice over IP Service changes its status to out-of-service or in-service respectively.
  
  Additionally, SIP Server now supports active out-of-service detection functionality for a device of type Voice over IP Service with service-type set to softswitch. The active out of service detection corresponds to the configuration when the option oss-check is not 0.
  
  The following log messages have been added:
  
  52000|STANDARD|GCTI_DEVICE_OUT_OF_SERVICE|Device [the name of the device] is out of service
  Description: The specified device is out of service based on active out-of-service detection.
  
  52001|STANDARD|GCTI_DEVICE_BACK_IN_SERVICE|Device [the name of the device] is back in service
  Description: The specified device is back in service based on active out-of-service detection.
  
  

  (ER# 197181299)

• SIP Server now provides the ability for parties participating in a consultation call to initiate 3pcc (third-party call control) transfers or conferences. A typical supported scenario would be:
  
  
  1. An inbound call is routed to Agent A.
  2. Agent A originates a consultation call with Agent B.
  3. Agent B originates a consultation call with Agent C.
  
  (ER# 205959541)



• SIP Server now provides a busy tone in 3pcc (third-party call control) scenarios in which calls are made to a busy external destination. For this feature to work, the sip-busy-type configuration option must be set to 0 (the default). (ER# 197149960)



• SIP Server now provides a busy tone in single-step transfer or by using the TRouteCall request scenarios in which calls are routed to a busy destination. For this feature to work, the sip-busy-type configuration option must be set to 0 (the default) and the ring-tone-on-make-call option must be set to true. (ER# 197181321)



• SIP Server now supports a new DN-level option, cpn.
  
  cpn
  Default Value: None
  Valid Values: The SIP URI format according to the Augmented Backus-Naur Form (ABNF) in RFC 3261
  Changes Take Effect: Immediately
  
  When the value is specified for a DN of type Trunk, it will be used as a user part of the SIP URI in the From header of the INVITE message sent by SIP Server through this trunk. This option must not be configured on trunks that are allocated for direct signaling between SIP Servers.
  
  Note: If the CPNDigits parameter is specified in the Extensions attribute in TMakeCall, TMakePredictiveCall, TInitiateConference, or TInitiateTransfer requests, it takes precedence over the cpn option setting.
  
  (ER# 197181291)



• SIP Server now provides the ability for an agent to handle up to three 3pcc calls on his or her SIP endpoint. This feature supports the following operations: hold, retrieve, and alternate. (ER# 195916587)



• SIP Server now supports the ISCC/COF feature.
  
  To support this feature:
  
  
  • A new private header X-ISCC-CofId has been introduced.
  
  
  
  • A new Application-level configuration option, default-network-call-id-matching, has been added, in the extrouter section.
  
  
  default-network-call-id-matching
  Default Value: None
  Valid Values: sip
  Changes Take Effect: Immediately
  
  When this option is set to sip, SIP Server will use the content of the X-ISCC-CofId header for the ISCC/COF call matching.
  
  

  To activate this feature, set the following options in the SIP Server's extrouter section:

  cof-feature=true
default-network-call-id-matching=sip

  (ER# 197181327)

• SIP Server now supports 3pcc (third-party call control) calls made to an ACD Queue. (ER# 197181295)



• SIP Server now supports the SIP Server-specific configuration option, default-dn, at a Switch/DN level:
  
  default-dn
  Default Value: NULL
  Valid Values: Any valid DN
  Changes Take Effect: Immediately
  
  This option specifies the DN to which calls are sent when URS is nonoperational, or when the timeout specified in the router-timeout option expires. This option does not apply to calls that are delivered to an ACD Queue associated with the Routing Point. This option can be configured only on DNs of type Routing Point.
  
  Note: The option settings at the Switch/DN level take precedence over the Application level settings.
  
  
• SIP Server can now send two types of DTMF tones: audio tones and named telephone events.
  
  To support this functionality, a new Application-level configuration option, sip-dtmf-send-rtp, has been introduced.
  
  sip-dtmf-send-rtp
  Default Value: false
  Valid Values: true, false
  Changes Take Effect: Immediately
  
  This option specifies whether SIP Server instructs Stream Manager to send DTMF tones when a T-Library client issues a TSendDTMF request. When this option is set to true, SIP Server instructs Stream Manager to send DTMF tones to all call participants using one or both of the following DTMF generation methods: RTP packets with Named Telephone Event (NTE) payload as specified by RFC 2833, and in-band audio tones according to ITU-T Recommendation Q.23. (ER# 175871109)



• SIP Server can now distribute information about all call participants—except the trunks allocated for communication between SIP Servers, distribution devices (such as Routing Points or ACD), and supervisors' DNs—to logged-in agents by using the SIP NOTIFY method and EventUserEvent messages.
  
  The information about the call participants is reported in the Extensions attribute of the relevant event using the following key-value pairs:
  
  Key: LCTPartiesLength
  Type: Integer
  Description: Specifies how many parties are involved in a single call.
  
  Key: LCTParty<n>
  Type: Integer
  Description: Represents a party of the call, where n is an integer value starting from 0.
  
  To support this functionality, a new Application-level configuration option, sip-enable-call-info, has been added.
  
  sip-enable-call-info
  Default Value: false
  Valid Values: true, false
  Changes Take Effect: Immediately
  
  

  When set to true, SIP Server distributes the information about call participants to logged-in agents by using the SIP NOTIFY method and EventUserEvent messages.

  To configure this functionality:

  1. Set the sip-server-inter-trunk options for DNs of type Trunk that are allocated for direct signaling between SIP Servers. The NOTIFY method will be sent only to sessions that are established through such trunks.
  
  
  2. Set the Application-level sip-enable-call-info configuration option to true.

  (FR# 198800873)

• SIP Server now supports the alternate call operation between the main call and consultation call that is queued on a Routing Point, as described in the following scenario:
  
  
1. A call is routed to Agent A.
2. Agent A places the call on hold and initiates a consultation call by dialing to a Routing Point.
3. Agent A is placed in a queue at the Routing Point, waiting for another agent to become available (a treatment is played).
4. Agent A places the consultation call on hold and retrieves the main call from hold.

For the alternate call operation to work transparently in a multi-site environment, a treatment must be applied to a call on a Routing Point at the earliest possible time. If a treatment is not applied, the alternate call operation will not be successful and SIP Server will generate an EventError (Call in invalid state) message.

(FR# 189387717)

• SIP Server now supports trunk optimization for multi-site transfer scenarios as described below.
  
  Scenario 1:
  
  1. An inbound call is routed to Agent A at the SIP Server 2 site.
  2. Agent A initiates a two-step transfer to Agent B at the SIP Server 1 site.
  In this scenario, SIP Server uses a SIP REFER request with the Replaces header to report call data for Agent B.
  
  Scenario 2:
  
  1. An inbound call is routed to Agent A at the SIP Server 1 site.
  2. Agent A initiates a two-step transfer to Agent B at the SIP Server 2 site.
  In this scenario, SIP Server uses a SIP INVITE request with the Replaces header to report call data for Agent B.
  
  Scenario 3:
  
  1. From a SIP Server 1 site, a call arrives to Agent B at the SIP Server 2 site.
  2. Agent B initiates a two-step transfer to Agent C at the SIP Server 3 site.
  In this scenario, SIP Server uses a SIP INVITE request with the Replaces header to report call data for Agent C.
  
  When the trunk optimization functionality is in use, the OtherDN attribute contains correct information and is reported properly in EventPartyChanged messages in the above scenarios.
  
  To support this functionality, a new DN-level configuration option, sip-server-inter-trunk, has been added.
  
  sip-server-inter-trunk
  Default Value: false
  Valid Values: true, false
  Changes Take Effect: Immediately
  
  When set to true, depending on the scenario, SIP Server determines whether to complete the transfer operation using the REFER or INVITE with the Replaces header.
  
  To configure this functionality:
  
  
  1. In each SIP Server configuration (origination and destination), in the corresponding SIP Switch, configure a DN of type Trunk. These Trunk DNs will be used for direct signaling between SIP Servers.
  
  
  
  2. Set these configuration options as follows:
     
     sip-server-inter-trunk=true
refer-enabled=true
oosp-transfer-enabled=true
     
     
  3. In multi-site routing, to avoid reporting an access resource as AttributeOtherDN in related events, Genesys recommends using ISCC direct transaction types (such as direct-uui). See the Multi-Site Support chapter of the SIP Server Deployment Guide for configuration details.

  (FR# 192436073)

• SIP Server now allows an agent to complete a consultation transfer of a call that is located on a Routing Point. This transfer operation is supported in single-site and multi-site environments.

  In a single-site environment, the call transfer can be completed both when a treatment is playing for the call on a Routing Point, or when the call is just parked on a Routing Point.

  In a multi-site environment, when a consultation call is made to a Routing Point located on another site, the call transfer can be completed only when a treatment is playing for the call on the Routing Point. If a call is just parked on a Routing Point, the complete transfer operation will not be successful and SIP Server will generate an EventError (Call in invalid state) message.

(FR# 192133611)

• SIP Server can now provide a silent treatment for conference call participants when one of them places the call on hold. This will allow conference call participants to continue the conference without interruption (or hearing the music-on-hold treatment). This feature is applicable to conference calls where participants are located in single-site or multi-site environments.

  To support this functionality, a new Application-level configuration option, music-in-conference-file, has been added.

  music-in-conference-file
  Default Value: The value is taken from the default-music option
  Valid Values: A string containing the valid name of the music file
  Changes Take Effect: Available for next 3pcc or 1pcc hold operation

  Specifies the silent audio file to be played in applicable conferences (more than two active participants). If the conference has only two active participants, then the music file defined in the default-music option will be played for the other party when the call is placed on hold. For conferences with more than two active participants, the music-in-conference-file option is used for a silent MOH treatment instead. Recommended value is music/silence.

  For example, if a supervisor in silent monitoring mode listens in on a call between a customer and an agent, the supervisor is not considered an active participant. If the agent or customer places the call on hold, the remaining participants will hear the default-music MOH treatment. However if the supervisor places the call on hold, the music/silence file is played instead, so that the customer and agent can continue their conversation undisturbed.

(FR# 179122061, 191804785)

• SIP Server now supports Audiocodes 2000 (v.5.20A.048.004) as an MCU (Multipoint Conference Unit).

Corrections and Modifications

This release also includes the following corrections and modifications:

In a multi-site environment, when a call is routed from one site to another site, SIP Server no longer applies a greeting on the site where a Routing Point is configured with the ISCC transaction type route. Previously, the greetings started simultaneously on both sites, which resulted in race conditions in signaling. For this scenario to work properly the sip-server-inter-trunk option must be set to true on trunks that are configured between SIP Servers. (ER# 212502181) See Known Issues and Recommendations: ER# 212618872, 212618863.

This version of SIP Server enables propagation of "rejected" responses back to a calling party. Previously, a busy tone was always applied and a calling party dialog was accepted for five seconds, which resulted in unnecessary charges to the calling party.

A new Switch/DN level configuration option (for DNs of type Trunk) controls the new SIP Server behavior:

sip-busy-type
Default Value: 0
Valid Values: 0, 1, 2
Changes Take Effect: Immediately

When this option is set to 0 (the default), a busy tone is always played. When this option is set to 1, a busy tone is played for a calling party only if a treatment is previously applied to a call or a call is originated by a 3pcc make call operation, and the refer-enabled option is set to false. Otherwise, the rejected response is sent back to the calling party. When this option is set to 2, a busy tone is not applied, and if SIP Server does not accept an INVITE session from a calling party, the rejected response is sent back to the calling party.

(ER# 208104341)

SIP Server now properly applies the reinvite-requires-hold configuration option for a conference completion. Previously, this would cause a delay in the media path with a SIP endpoint. (ER# 204773474, 209903868)

In multi-site supervision scenarios, SIP Server now correctly processes TSetMuteOn and TSetMuteOff requests from a remote supervisor with a connection ID that is different from the original call. (ER# 210272383)

SIP Server now generates an EventDialing message in a 3pcc (third-party call control) make call scenario, in which an INVITE request is sent to the endpoint. Previously, SIP Server generated EventDialing upon receiving a 200 OK message in response to the INVITE request, and this resulted in a usability issue on the agent desktop. (ER# 209372222)

SIP Server no longer adds additional quotation marks to the Display Name in the From header. Previously, some SIP endpoints could reject such an INVITE message containing double quotation marks in the From header. (ER# 206608077)

SIP Server now properly handles a scenario in which an internal party releases a call while the call is routed from a Routing Point to the destination. Previously in this scenario, SIP Server did not release the call, and mistakenly invited the internal party back to the treatment on the Routing Point. (ER# 206644231)

SIP Server no longer places a DN of type Trunk in the out-of-service state when a timeout is expired or a 408 Request Timeout response is received from a gateway in a SIP dialog that is associated with this Trunk DN. (ER# 208103312)

SIP Server now properly handles a configuration issue when an out-rule-<n> option is set to an empty value in the Class of Service feature configuration. Previously, SIP Server became unstable. (ER# 208271970)

SIP Server now correctly handles a multi-site call supervision session with the ISCC transaction type route. Previously, SIP Server mistakenly dropped the monitored call. (ER# 208690137)

SIP Server now properly handles a race-condition scenario in which a calling party does not respond to the active INVITE message, but releases the call (by sending BYE) before the 18x response is received from the routing destination. Previously in this scenario, SIP Server became unstable. (ER# 209035095)

SIP Server now properly applies the reinvite-requires-hold configuration option for a conference completion. Previously, this would cause a delay in the media path with a SIP endpoint. (ER# 204773474)

SIP Server now properly handles a scenario where a subsequent call treatment is rejected by Stream Manager because of configuration errors. Previously, SIP Server dropped the call at the Routing Point. (ER# 206109242)

SIP Server now reports a monitoring mode and a monitoring scope in the Extensions attribute in an EventEstablished message for a supervisor's DN. Previously, SIP Server reported monitoring parameters only in an EventRinging message. (ER# 206785614)

SIP Server now correctly processes TRequests received from an agent's T-Library client if a personal greeting is playing to the agent. (ER# 89217949, 170650597)

SIP Server now adds monitoring extensions in an EventRinging message for a supervisor's DN and correctly reports the current supervision session state, such as MonitorMode set to coach, mute (normal), or connect, and MonitorScope set to call or agent. (ER# 203483403)

SIP Server now properly processes a manual completion (when a phone sends a REFER message) of a call transfer to an external number, even if that number coincides with the Agent ID of a logged-in agent. Previously, SIP Server delivered calls to the DN at which the agent was logged in. (ER# 102069454)

SIP Server now properly processes a manual completion (when a phone sends a REFER message) of a two-step transfer. Previously, SIP Server could produce a memory access violation during the call release. (ER# 174995525)

SIP Server now forwards SDP information to the call origination DN in the 18x response message in first-party call control scenarios, where the ring-tone-on-make-call configuration option is set to false in the origination DN configuration. (ER# 198836221)

SIP Server now correctly handles provisional responses from media gateways containing early media, when outbound calls are made using TMakeCall requests and a call origination device has the make-call-rfc3725-flow configuration option set to 1 in its configuration. After receiving an 18x response containing SDP information from the destination, SIP Server provides the received early media to the call origination device when the ring-tone-on-make-call option is set to false in the device configuration. This enables the RTP session to be properly established between media gateways and the call origination device. Previously, SIP Server ignored early media received from the gateway, and responded to the call origination device only after receiving the 200 OK response from the gateway. (ER# 198297652)

SIP Server now correctly processes the SDP information with multiple a= headers that do not belong to any media (m=) section (for example, a= headers are located before the first m= header in the SDP). Previously, SIP Server failed to process such an SDP. (ER# 199747389)

When operating in a multi-site environment, SIP Server now correctly handles race conditions that occur when a personal greeting is applied to a call. Previously, the call was dropped because SIP Server generated a 488 (Not Acceptable Here) response message. (ER# 201312162)

Personal greeting functionality is now available for predictive calls. (ER# 112238974)

SIP Server now starts call recording in a scenario where a call, after a voice treatment completion, is transferred to an agent endpoint, and the agent endpoint responds to the INVITE message with a 200 OK message that contains an SDP in which both a video codec and an audio codec are present. Previously, SIP Server did not start call recording if the video codec was present in such a response. The issue applied to scenarios where the record option was set to true in the agent DN configuration. (ER# 194729141, 174433788)

In a scenario involving a TMakePredictiveCall request, where a gateway responds to the INVITE request with a 480 error message, SIP Server now distributes AttributeCallState set to 6 (Busy) in the EventReleased message. Previously in this scenario, SIP Server distributed AttributeCallState set to 0 (OK). (ER# 193633122)

When propagating messages between two connected endpoints, SIP Server no longer places two Privacy headers inside the outgoing INVITE message if the incoming INVITE message contains only a single Privacy header. This issue occurred when SIP Server communicated with endpoints via the ACME server and SIP Server received a 400 Multiple Privacy headers error message from the ACME side. (ER# 147873387)

SIP Server now properly distributes an EventEstablished message when processing a TMakePredictiveCall request made to an internal DN. Previously, SIP Server did not distribute such an event message. (ER# 121384322)

SIP Server now enables an agent to send a TAgentLogout request and generates an EventAgentLogout message as a response when a 3pcc call initiated by the agent is in progress (the destination party is in the Ringing state). Previously, SIP Server generated an EventError message for the dialing party in response to the TAgentLogout request, and, if the agent's desktop disconnected, SIP Server generated EventAgentLogout with ReferenceID. (ER# 156761975)

When processing a TCompleteTransfer request using the REFER method, SIP Server now terminates a dialog with a transferred party that accepts REFER. Previously, SIP Server did not terminate such a dialog, which caused some devices to remain on hold even after the transfer completed. (ER# 156024561)

When a call is routed to a destination that responds with a 603 Decline message, SIP Server now issues a DN is Busy error message. Previously, SIP Server did not send this error message, and the call was forwarded to the DN specified by the value of the default-dn configuration option immediately after receiving the 603 Decline response. In some cases (if a DN of type ACD Queue was specified as the value of the default-dn option), this led to stuck calls. (ER# 164964266)

SIP Server no longer distributes an EventError message when it receives a TDeleteFromConference request for a conference participant that is in a Ringing state. Previously, SIP Server incorrectly distributed such an event message. (ER# 167728850)

When a call is routed to an agent phone that is configured behind a softswitch and several softswitches are used in the configuration, SIP Server now correctly performs load balancing among the softswitches. Previously, SIP Server did not always perform load balancing correctly. (ER# 179787375)

SIP Server now correctly terminates all SIP transactions associated with scenarios where an agent, whose device has a record option set to true, sends a TMakeCall request, and then sends a TReleaseCall request while the SIP transaction with the recording device is not fully completed. Previously, when the call was released, SIP Server did not terminate the SIP transaction with the recording device. (ER# 193480857)

After a switchover, SIP Server now correctly responds to a TRegisterAddress or TUnregisterAddress request that is invoked on behalf of a DN of type Communication DN. Previously, SIP Server sometimes rejected such a request with an error message (DN is not configured in CME). (ER# 188076203)

When a caller sends a BYE message before the routing destination can answer, SIP Server now correctly sends a CANCEL message and terminates the dialog with the routing destination. Previously in this scenario, SIP Server sent a BYE message instead. (ER# 187727745)

SIP Server no longer sends both EventBackInService and EventOutOfService messages in addition to the EventOutOfService message if a DN object is marked as Disabled in Configuration Manager. (ER# 184633347)

SIP Server can now retrieve a call in the following scenario:

1. The call is being recorded by Stream Manager.
2. The call is placed on hold.
3. Stream Manager is stopped.

When GVP operates in stand-alone mode and when IPCS is configured as Trunk, SIP Server now creates a subscription based on the SUBSCRIBE request received from GVP Resource Manager (RM). (ER# 193876231)

In the following scenario, where Outbound Solution operates in HMP-ASM mode:

1. An outbound call is initiated by CPD Server using the SIP protocol.
2. A destination party responds with a 486 Busy Here message.

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Supported Operating Systems

New in This Release

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 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 the initial 7.6 release of SIP Server.

• Enhanced instant messaging support. SIP Server now supports the following call supervision modes for instant messaging (IM) calls: Silent Monitoring, Whisper Coaching, Open Supervisor Presence, and Intrusion. SIP Server also supports multiple IM sessions and delivery of Instant Messages transcripts.

• Class of Service support. Class of Service is the functionality that defines telephony capabilities for a device or an agent.

• Remote Supervision support. The Remote Supervision feature enables supervisors to monitor agent calls from outside the contact center.

• Multi-Site Supervision support. The Multi-Site Supervision feature enables supervisors at a local site to monitor agents located at remote sites.

• Enhanced SIP headers mapping support. SIP Server can now extract data from a REFER message, and map it to either the Extension or UserData attribute of T-Library event messages.

• Enhanced single-step transfer to a Routing Point support. SIP Server can now perform a single-step call transfer to a Routing Point DN when the call is silently monitored by a supervisor, or when the call is in emergency recording mode.

• TAlternateCall request support. SIP Server now supports TAlternateCall requests for DNs configured with the dual-dialog-enabled configuration option set to either false or true.

• Trunk capacity configuration. The number of calls per trunk can now be controlled with new DN-level configuration options: capacity and capacity group.

• Stream Manager overloading scenario support. SIP Server supports the load-sharing mechanism implemented in Stream Manager. In an environment with distributed Stream Managers, if a dialog is rejected solely because the rejecting Stream Manager is overloaded, SIP Server tries to re-send that dialog to another Stream Manager. See the Framework 7.6 Stream Manager Deployment Guide for details.

• OPTIONS messages support. SIP Server now supports processing of OPTIONS messages.

• DTMF tones generation support. SIP Server can now send a request to Stream Manager to generate DTMF tones using the TApplyTreatment request with TreatmentType set to PlayApplication.

• Enhanced high-availability support. When operating in a high-availability environment, SIP Server can now synchronize calls that are in ringing state.

Corrections and Modifications

This release includes the following corrections and modifications that were made between release 7.5 or earlier releases and the initial 7.6 release:

SIP Server now correctly processes the following scenario:

1. An inbound call is delivered to an agent.
2. The agent uses a first-party call control (1pcc) blind transfer to transfer the call to another agent.
3. The transfer destination does not respond to the INVITE message from SIP Server.
4. The external caller drops the call.

SIP Server now correctly processes the following scenario:

1. An inbound call arrives at a Routing Point.
2. The call is routed to DN1.
3. The call is transferred using a first-party call control (1pcc) blind transfer to DN2.
4. The call is answered at DN2.
5. The call is transferred to an unavailable external destination (the 404 Not Found message is generated) using the 1pcc blind transfer.
6. At DN2, the call is released using the T-Library request.

SIP Server now properly generates an EventRinging message in a scenario where a call is routed and redirected multiple times before it is delivered to an agent. (ER# 163257279)

SIP Server now properly handles a scenario where no response is received to the INVITE message for the treatment service. Previously in this scenario, SIP Server could become unstable. (ER# 163257285)

SIP Server now correctly selects a trunk when a find-trunk-by-location configuration option is set to true. (ER# 163257301)

SIP Server no longer becomes unstable when a music-on-hold service fails during a Hold operation. (ER# 163257331)

SIP Server now correctly releases unanswered consultation calls. Previously in some scenarios, SIP Server did not release such calls, which resulted in reporting incorrect agent states. (ER# 163257343)

SIP Server now correctly releases a main call during race conditions, when a caller releases the call and an agent attempts to complete the transfer at the same time. Previously in this scenario, the main call could became stuck. (ER# 163257381)

For an inbound call that is being recorded, SIP Server now sends the re-INVITE request to the external call party when the session-refresh-interval timer expires. (ER# 183499931)

The Genesys call model is now fully followed in a scenario where a 1pcc (first-party call control) blind transfer is made to a Routing Point within a consultation call that is created during another two-step transfer. (ER# 179504371)

In a scenario where an inbound call from DN 1 (SIP Server 1) is established to DN 2 (SIP Server 2), and an agent at DN 2 initiates a transfer to DN 3 (SIP Server 1), SIP Server 1 now distributes TEvents for DN 3. (ER# 177935631)

SIP Server no longer starts processing a TAlternateCall request for a call, even if the call's ConnID attribute is missing in the request. (ER# 168729706)

If a TSendDTMF request is issued while a destination party is in the ringing state, SIP Server no longer sends the INFO requests to the destination party after the destination party has answered the call. This issue arose when the destination party was configured with the rfc-2976-dtmf configuration option set to true. (ER# 164122477)

If a TSendDTMF request is issued and SIP Server receives a 405 Method Not Allowed message as a response to the INFO request, SIP Server now reports an error. Previously, SIP Server sent an EventDTMFSent message instead. (ER# 164682262)

SIP Server now correctly processes a TSendDTMF request made during a consultation call if a conference or transfer is performed to an internal party. SIP Server also sends INFO requests to the internal party after the internal party issues the TCompleteConference or TCompleteTransfer, and TSendDTMF requests. This issue applies when the internal party was configured with the rfc-2976-dtmf configuration option set to true. (ER# 164391943)

The ringback tone can now be played during a single-step transfer or a single-step conference of an internal call under the following conditions:

1. The 180 Ringing message is received from the destination.
2. The option ring-tone-on-make-call is set to true.
3. The option refer-enabled is set to false.

SIP Server no longer rejects a TMonitorNextCall request with an EventError message if a personal greeting is being played. (ER# 111800779)

SIP Server now generates an EventPartyDeleted message for the monitoring supervisor's desktop when a non-monitoring party releases the call. (ER# 88156295)

SIP Server now correctly generates an EventAbandoned message in a scenario where a single-step call transfer is performed to a Routing Point using the REFER method and the transferred party releases the call before routing is completed. (ER# 82389690)

If a supervisor issues a TMonitorNextCall request while a personal greeting is active, SIP Server no longer responds with an EventError message. (ER# 111800779)

A party on a consultation call is no longer put on hold when a switchover occurs in high-availability mode while Stream Manager is not operating. (ER# 108020587)

Video is now available when a TMakeCall request is issued for an EyeBeam phone and the ring-tone-on-make-call option is set to true. (ER# 111137259)

An agent's personal greeting is now performed when an inbound call to an ACD Queue associated with a Routing Point is routed to an agent. (ER# 113847720)

A call is now correctly reconnected if the TReconnectCall request follows immediately after the EventEstablished message. (ER# 116659592)

If the divert-on-ringing option is set to false, SIP Server now correctly distributes an EventAbandoned message when the destination DN responds with an error. Previously in this scenario, SIP Server distributed an EventRouteUsed message. (ER# 112941292)

SIP Server now correctly retrieves a call from the Hold state when a Voice over IP Service DN with the service-type option set to music is not responding. (ER# 114536401)

In environments containing EyeBeam endpoints with an Asterisk switch, TInitiateTransfer or TInitiateConference requests now work properly. (ER# 112026334)

SIP Server no longer continues to use a Trunk Group DN object after it has been deleted in Configuration Manager. (ER# 64735217)

Agents that are controlled by T-Server for Avaya Communication Manager can now perform a mute call transfer from T-Server back to GVP when the deployment contains an Avaya S8X00 switch, and SIP Server/GVP are configured in Behind-the-Switch mode. (ER# 72322301)

SIP Server now correctly updates the Contact option for a DN object in Disabled status in Configuration Manager when the REGISTER message is received from a SIP endpoint. (ER# 75353631)

SIP Server now generates an EventEstablished message if it receives an OK response message from an INVITE request before receiving an OK response message from a PRACK request. (ER# 64151723)

When SIP endpoints contain the dual-dialog-enabled option set to false, SIP Server no longer distributes an extra EventEstablished message for a consultation call even if the consultation call has not been answered. Previously, this occurred when an inbound or internal call was processed through a Routing Point, answered by an agent, and then transferred using a third-party conference control (3pcc) two-step transfer to an external destination. (ER# 63796621)

When a single-step or two-step transfer is completed between site A and site B, it is now possible for site B to transfer the call back to site A. (ER#s 58409511, 58099611)

When a destination DN is configured with the dual-dialog-enabled option set to false, a single-step call transfer to a Routing Point that is then routed to a SIP endpoint no longer fails. (ER# 54944121)

When the dual-dialog-enabled option is set to false, the ringing tone now stops playing when the consultation call has been released. (ER# 54579795)

When the dual-dialog-enabled option is set to false, a busy tone is now generated if the consultation call is made to a busy destination. (ER# 54631261)

SIP Server no longer reports the EventDNOutOfService message for a DN that an agent is logged into after a call that was delivered though ACD (using emergency routing) is abandoned. (ER# 32468545)

SIP Server now always updates the contact field in the Annex tab when contact information was changed in the REGISTER request or the value of the internal-registrar-persistent option is changed to true. (ER#s 30340003, 28788635)

SIP Server no longer exits unexpectedly while attempting to retrieve a primary call from an H.323 endpoint after a 1pcc consultation call has been released by an H.323 endpoint. (ER# 37915522)

SIP Server no longer exits unexpectedly while attempting to execute a RequestClearCall on a consultation call after an unsuccessful attempt to complete a transfer. (ER# 26984886)

SIP Server no longer reports an incorrect call state in the EventPartyDeleted message when one of the participants is removed from the conference as a result of the TReleaseCall request. (ER# 25600073)

When a DN is deleted from Configuration Manager, SIP Server sends an EventUnregistered message to respective clients and continues to work with this DN as an external device. (ER# 29864629)

A first-party call control (1pcc) transfer by re-INVITE from a Grandstream phone is now supported. (ER# 27442896)

A 1pcc single-step transfer to a Routing Point from Grandstream/Zultys phones no longer produces an incorrect call state in EventRinging and EventQueued messages. The call state is also reported in the EventPartyChanged message. (ER#s 27442926, 27442922)

DN in-service and out-of-service states are now correctly propagated from the primary SIP Server to the backup SIP Server. (ER# 11288354)

SIP Server no longer distributes the EventPartyChanged message on the remote site when a transfer/single-step transfer is completed on the local site. For conferences, SIP Server distributes EventPartyChanged or EventPartyAdded. (ER# 98921)

SIP Server can now process TAgentLogout or TSetNotReady requests for an agent with an active call. Previously, SIP Server generated an error. (ER# 96236)

SIP Server no longer produces a memory access violation when a DN, on which a client is registered, is deleted from the Configuration Layer. Previously, this error only occurred when the Application-level option, dn-del-mode, was set to idle. (ER# 122731143)

SIP Server now gracefully handles a greeting failure. Previously, SIP Server would disconnect a call if a greeting failed for any reason (for example, the announcement file was missing). (ER# 182662698)

SIP Server no longer ignores the find-trunk-by-location and geo-location options in certain 1pcc outbound call scenarios. Previously, ignoring these options sometimes resulted in the wrong gateway being selected for the outbound call. (ER# 100864216)

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Known Issues and Recommendations

This section provides the latest information on known issues and recommendations associated with this product.

The Framework 7.6 SIP Server Deployment Guide contains misleading information about the HA script sample in the "Recommended Sample Batch File" section of the High-Availability Deployment chapter. The information from the "Recommended Sample Batch File" section must not be used as reference for building NLB cluster control scripts. For complete information about the recommended SIP Server HA deployment, refer to the white paper SIP Server 7.6 – HA Configuration, available on the Customer Care website. (ER# 241796966)

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Discontinued Support

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 RTC-based phones that are registered to AcmePacket SBC (Session Border Controller) (version ACME Firmware 4.1.1 Patch 64).

SIP Server no longer supports the confirm-rtp configuration option that was introduced in 7.5 release. SIP Server now applies the same logic as it had done when the confirm-rtp option was set to true. (ER# 240349863)

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Internationalization

Information in this section is included for international customers.

There are no internationalization issues for this product.

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Additional Information

Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. The following documentation also contains information about this software.

• Framework 7.6 SIP Server Deployment Guide contains detailed reference information for the Genesys Framework 7.6 SIP Server, including configuration options and specific functionality.

• Framework 7.6 Deployment Guide helps you configure, install, start, and stop Framework components.

• Genesys Events and Models Reference Manual contains the T-Library API, information on TEvents, and an extensive collection of call models.

• Genesys Migration Guide contains a documented migration strategy for each software release. Please refer to the applicable portion of this guide or contact Genesys Customer Care for additional information.

• 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) or the Developer Documentation CD.

Note: For the DVD/CD, the New Documents on this DVD/CD 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.

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