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SIP-I, Session Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. RFC 3261 SIP: Session Initiation Protocol June 2002 The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. Figure 1 shows a typical example of a SIP message exchange between two.
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Sip 1 1 5 Equals
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This product includes software developed by the Apache Software Foundation (http://www.apache.org/).
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Licensing:
Licensing section of the Genesys Migration Guide
Information on supported hardware and third-party software is here:
SIP Server 8.1 Product AlertsIn the 8.1.1 release, SIP Server and SIP Proxy implemented the operational statistic query through Local Control Agent (LCA) functionality (see the operational-stat-timeout option description in the SIP Server Deployment Guide). This functionality requires Management Framework 8.1.2 or later. 
On the Windows platform, LCA sometimes did not respond to requests from Solution Control Server because LCA was waiting for a response to its Windows Management Instrumentation (WMI) query for performance data. 
Starting with LCA version 8.5.100.09, the WMI response time interval can be configured for LCA to provide timely handling of requests from Solution Control Server. See release notes for LCA version 8.5.100.09 for details on the LCA configuration option.
New in Release 8.1.1Introduced in 8.1.104.00Support for AudioCodes 440HD, 430HD, 420HD, and 405 IP Phones with firmware version 2.2.16.428.
Introduced in 8.1.103.98Support for the Microsoft Windows Server 2019 operating system.
Introduced in 8.1.103.95For the HTTP Monitoring Interface feature, SIP Server adds monitoring of state and quantitative statistics for T-Library client connections of the following SIP Server threads: Session Controller, T-Controller, Interaction Proxy, and Smart Proxy. This feature is enabled by the new t-library-stats-enabled configuration option.
SIP Server supports CounterPath Bria v6.1.0, including audio only.
Introduced in 8.1.103.94SIP Server in Cluster mode can now optimize Smart Proxy logs (1792) to print T-Library events that are sent to or received from Smart Proxy clients in a summarized format. To enable this feature, set the new restricted smart-proxy-verbose-level option on the SIP Cluster Node DN (service-type=sip-cluster-nodes).
Introduced in 8.1.103.92To extend functionality that was introduced in release 8.1.103.66, SIP Server in standalone mode now supports passing the additional GVP parameters (which have Agent Assist supporting key-value pairs (KVPs) and Streaming KVPs) from AttributeExtensions of TRouteCall to MCP in the recording INFO messages, under existing recording metadata.
Introduced in 8.1.103.88SIP Server can now unmask specific SIP headers contained in SIP Server logs. The feature is enabled by the new x-sip-unmask-sip-headers and x-sip-unmask-headers-default configuration options in the [log] section of the SIP Server Application. See Masking sensitive data in SIP messages for details.
Introduced in 8.1.103.87SIP Server can now resolve a Fully Qualified Domain Name (FQDN) specified in the contact option of a DN using the asynchronous DNS resolution method and place the DN out of service if the FQDN is unresolvable. The feature applies to DNs of type Extension, ACD Position, and Voice Treatment Port. The DN-level enable-async-fqdn-resolve configuration option enables this feature.
Introduced in 8.1.103.85To extend functionality that was introduced for General Data Protection Regulation (GDPR) scenarios in releases 8.1.103.28 and 8.1.103.78, SIP Server now supports passing AttributeExtensions of a TRouteCall request into AttributeExtensions of an internal TMakePredictiveCall request when an agent makes a manual outbound call.
Introduced in 8.1.103.80SIP Server can now handle different HTTP error responses from SIP Feature Server for Dial Plan extended service (XS) requests in an enhanced way to address connection instabilities and provide a quality response to the origination side. Several new configuration options are added:enable-enhanced-dialplan-handling
xs-request-timeout
xs-post-timeout
xs-heartbeat-timeout
xs-missed-heartbeat-threshold
switchover-on-xs-oos
time-before-switchover-on-xs-oos
xs-pool-size
xs-heartbeat-interval
Introduced in 8.1.103.78To extend functionality that was introduced for General Data Protection Regulation (GDPR) scenarios in release 8.1.103.28, SIP Server now supports passing AttributeExtensions of a TMakeCall (with CPD) request into AttributeExtensions of an internal TMakePredictiveCall request and only those AttributeExtensions that are supported by the TMakePredictiveCall request. The new DN-level configuration option, make-call-cpd-extensions, is added.
Enhanced disaster recovery solution for outbound calls: After receiving a negative response, SIP Server can now select an alternative trunk for outbound calls. In addition, SIP Server can now attempt to connect to a DN via an alternative softswitch (found in the DN configuration) if the first attempt to connect to a DN via a softswitch resulted in a negative response from that softswitch. The new Application-level configuration option sip-error-codes-overflow is added.
Introduced in 8.1.103.75Support for Avaya Session Manager 8.x.
Introduced in 8.1.103.74To extend functionality that was introduced in release 8.1.103.35, SIP Server can now treat 1pcc calls from any Resource Manager as internal.
Introduced in 8.1.103.73In a scenario where a remote agent located behind a PSTN trunk places a call on hold, SIP Server can now connect the on-hold party to a media service with a silent treatment to prevent disconnection of the call by the trunk.
Introduced in 8.1.103.72For the HTTP Monitoring Interface feature, SIP Server adds the following statistics in separate tables for Trunk, Softswitch, MSML, and Trunk Group devices: error statistics, the total number of calls created on the device, the number of out-of-service detection instances per device, and location matching instances.
Introduced in 8.1.103.68Per compliance with RFC 3264, SIP Server now supports multiple media (m=) lines in SDP headers for interchangeable RTP/SRTP support.
Sp 1150-03Introduced in 8.1.103.66SIP Server in SIP Cluster mode now supports passing the additional GVP parameters (which have Agent Assist supporting KVPs and Streaming KVPs) from AttributeExtensions of TRouteCall to MCP in the recording INFO messages, under existing recording metadata. The new configuration option, record-metadata-prefix, is added.
Introduced in 8.1.103.65Starting with this release, SIP Server requires FlexNet Publisher License Manager version 11.13 or later.
Introduced in 8.1.103.64High-availability enhancement: The backup SIP Server can now check the status of its internal components during startup to ensure it provides the service when it is promoted to Primary. If any internal component (T-Controller, Smart Proxy, Interaction Proxy, or Operational Information thread) fails to complete initialization, SIP Server reports the SERVICE_UNAVAILABLE status to the Management Layer (LCA/SCS). In addition, if SIP Server fails to open a SIP port during startup, it reports the SERVICE_UNAVAILABLE status to LCA/SCS. Note that this approach does not work for the IP Address Takeover procedure or same host configuration. The following new configuration options are added: backup-init-check, backup-init-check-timeout, and backup-sip-port-check.
Introduced in 8.1.103.63SIP Server supports new configuration options: sip-progress-response-code and sip-retry-after.

Introduced in 8.1.103.62As per GDPR requirement, on an outbound call made by a TMakeCall request through a Routing Point, SIP Server supports playing an opt-out prompt to an end user, so that the user can decide whether to opt out from recording and monitoring. This prompt can now be played in the customer's preferred language.
Introduced in 8.1.103.61When sending an ACK message with an answer SDP before terminating an agent dialog, SIP Server now includes a codecs list matching the offer SDP that is received for the dialog.
SIP Server now supports enhanced high-availability resilience for network disruptions. The following new configuration options are added: switchover-on-msml-oos and switchover-on-trunks-oos.
Introduced in 8.1.103.59Per PCI compliance, SIP Server supports sending DTMF digits to an inbound leg of the call in SIP INFO requests.
Introduced in 8.1.103.58This release improves music-on-hold handling in deployments where SIP Server is used with Secure Real-Time Transport Protocol (SRTP). Music continues playing if an agent places a call on hold. To enable this functionality, set the sip-reinvite-action configuration option to after-hold on a Trunk DN where a re-INVITE message arrives.
To troubleshoot T-Controllers (TC) communication issues, new statistics, based on periodic T-Library polls between T-Controllers, are now available. The new configuration option, tc-latency-poll-interval, is added.
Introduced in 8.1.103.56SIP Server in SIP Cluster mode now enforces authenticated access from SIP devices. To enable this functionality, set the sip-enable-strict-auth option to true on the SIP Cluster Node VOIP Service DN. This option can also be set at the SIP Server Application or at DNs of type Extension.
Introduced in 8.1.103.54SIP Server now supports an enhanced high-availability procedure where it reports a value of 0 for 'Readiness for Switchover' in an overloaded condition (CPU usage is above the threshold).
SIP Server now supports enabling the Preview Interactions feature using the TRouteCall request containing the preview-interaction key in AttributeExtensions.
Introduced in 8.1.103.53SIP Server in SIP Cluster mode: A new Active Out-of-Service (OOS) check procedure for a VoIP service or Trunk DN ensures that this DN is placed out of service only when SIP Proxy processes the SIP traffic. The following new options are added: sipp-oos-recheck and switchover-on-sipp-oos.
Introduced in 8.1.103.52SIP Server in SIP Cluster mode adds support for remote agents to use external numbers that are not provisioned in the Configuration Database. A new configuration option, replace-agent-phone, is introduced.
SIP Server can now detect a T.38 fax SDP and drop the call when using TMakePredicitveCall requests with Call Progress Detection. For SIP Server to drop the call when T.38 fax is detected, set the fax-detected option to drop.
Introduced in 8.1.103.46In a scenario where an agent places a call on hold and SIP Server receives a hold INVITE message from a Trunk (containing the 'a=inactive' SDP), SIP Server can delay sending a re-INVITE message when connecting a Music-on-Hold (MOH) service.
Introduced in 8.1.103.45In SIP Cluster deployments, because of network issues, the connection between SIP Cluster nodes can be lost and promptly restored. T-Controller, the T-Library interface, maintains a subset of agent/DN states and interconnection to other T-Controllers in the cluster. A new configuration option, tc-reconnect-timeout, handles network outages gracefully in your environment.
In SIP Cluster deployments, where SIP-based DN ownership can be moved from one data center to another and, as a result, agent states are automatically changed to the logout state, SIP Server can now restore agent states immediately. A new configuration option, agent-state-auto-restore, handles agent states gracefully in your environment.
Introduced in 8.1.103.41Support for Polycom VVX phones (VVX 501, VVX 601), including video call support for Polycom EagleEye Mini Camera (5.9.1 and later). This support requires Media Control Platform (MCP) version 9.0.019.68 or later.
Enhanced support for integration with Unify OpenScape Voice using an alternative HA deployment. See SIP Server and OpenScape Voice Integration Overview for details.
Introduced in 8.1.103.40The following new configuration options are added (see the RN for details):sip-call-id-suffix
override-domain-ruri
sip-contact-user
sip-add-via
Introduced in 8.1.103.37For Windows and Linux operating systems, SIP Server now uses its own implementation to retrieve CPU and memory usage of SIP Server processes and threads. Previously, SIP Server used LCA to retrieve CPU and memory usage.
In SIP Cluster deployments, TEvents distribution by Session Controller to the Interaction Proxy (I-Proxy) running in HA mode is optimized.
Introduced in 8.1.103.35For Genesys Engage cloud deployments only: SIP Server can treat 1pcc calls from external callers (agents behind SIP trunks) as inbound calls. For enabling this feature, contact Product Management or your Genesys representative.
The external-contact option can now be changed dynamically. Previously, restart of SIP Server was required for the changes to take effect.
SIP Server supports Oracle Linux version 7.
For SIP Cluster mode, the default value of common/enable-async-dns is changed from 0 to 1.
When running in SIP Cluster mode, SIP Server now communicates with the SIP Feature Server dial plan through a new Extended Services component (that is run in its own thread) instead of the Call Manager thread.
When running in SIP Cluster mode, SIP Server can now terminate DN ownership in a T-Controller internal module by processing a TPrivateService request containing AttributePrivateMsgID=8273 for AttributeThisDN.
Introduced in 8.1.103.28SIP Server can now disable recording and monitoring of outbound calls in SIP Cluster deployments.
Introduced in 8.1.103.26SIP Server can now enable call recording on the agent side in SIP Cluster deployments.
Introduced in 8.1.103.25In standalone mode, SIP Server now starts a timer after sending an OPTIONS request through a SIP Proxy to a DN.
SIP Server adds the ability to monitor statistics related to SIP Feature Server interactions.
Introduced in 8.1.103.24Support for Oracle Enterprise Session Border Controller AP 1100/3900/4600/6300/6350 version E-CZ8.1.0.
Support for AudioCodes 440HD, 430HD, 420HD, and 405 IP Phones with firmware version 2.2.16.
When running in SIP Cluster mode, SIP Server now offers the T-Library interface, called Smart Proxy, for event monitoring clients (such as Stat Server).
Introduced in 8.1.103.20When running in SIP Cluster mode and when After Call Work (ACW) is controlled by UserData, SIP Server now applies the ACW state to an agent correctly.
Introduced in 8.1.103.16Support for F5 Networks BIG-IP Local Traffic Manager version 12.
Introduced in 8.1.103.10Support for Avaya Session Manager version 7.x.
Introduced in 8.1.103.08Support for secure SIP signaling, or sips, in accordance with RFC 5630.
Support for CentOS Linux 7.
Introduced in 8.1.103.06Support for a TLib-based Instant Messaging session initiated by an agent with the nailed-up connection.
Introduced in 8.1.103.02When MCP sends updated user data with an empty value in the MSML Info body, SIP Server now updates the user data with the empty value in T-Library events.
Support for a new Application-level configuration option sip-elin-timeout, for the integration with the RedSky emergency call provider.
Introduced in 8.1.102.99Support for CounterPath Bria v5.1.0, including audio and video.
Introduced in 8.1.102.98Support for AudioCodes 420HD IP Phone with firmware version 2.2.12.
Introduced in 8.1.102.95Support for a new Application-level configuration option report-error-on-routing-end, to generate EventError with ErrorCode 453 and the Call has been disconnected error message.
Synchronization of multisite (ISCC) calls that start before the HA connection between the primary and backup SIP Servers is established.
Support for Windows Server 2016 operating system. See the SIP Server page in the Genesys Supported Operating Environment Reference Guide for more detailed information and a list of all supported operating systems.
Introduced in 8.1.102.93Support for remote agents and agents with nailed-up connections to use external numbers that are not provisioned in the Configuration Database.
Support for AudioCodes 405HD and 440HD IP Phone with firmware version 2.2.12.
Introduced in 8.1.102.92Enhanced support to SIP Feature Server for assigning calling profiles (such as dial plans) to Routing Point and Trunk Group DNs for predictive calls made on behalf of a Routing Point or Trunk Group DN. This functionality is required for consistency with the internal SIP Server dial plan. A new Application-level configuration option, enable-outbound-ext-dial-plan, supports this feature. This feature depends on support from a specific version of SIP Feature Server. Consult SIP Feature Server documentation for the availability of this new feature.
Introduced in 8.1.102.89Support for providing a ringback tone to 1pcc calls. This behavior applies to inbound calls from external DNs.
Introduced in 8.1.102.81Support for the SIP Event Package for Registrations, in accordance with RFC 3680.
Support for a secure connection to Message Server using the Transport Layer Security protocol.
Introduced in 8.1.102.79Support for a new Application-level option unknown-gateway-reject-code to define which SIP error code SIP Server returns when an incoming INVITE message cannot be associated with an internal device or trunk, when the enable-unknown-gateway option is set to false.
Introduced in 8.1.102.73Support for masking of DTMF information in some attributes of T-Library messages of SIP Server log files.
Support for enhanced reporting of the monitoring mode in EventUserEvent messages to ICON that includes LCT parties information.
Sip 1 1 5 EqualsFor the features introduced in previous releases, see SIP Server 8.1.x CD Readme.
Genesys SIP Server is the Genesys software component that provides an interface between your telephony hardware and the rest of the Genesys software components in your enterprise. It translates and keeps track of events and requests that come from, and are sent to the telephony device. SIP Server is a TCP/IP-based server that can also act as a messaging interface between SIP Server clients. It is the critical point in allowing your Genesys solution to facilitate and track the contacts that flow through your enterprise. 
Fonepaw android data recovery crack. Genesys SIP Server is included in various packages of the Inbound, Outbound, and Self-services solutions for Customer Engagement and the Platform and Integration solutions for Business Optimization. Contact your Genesys representative for more information.
What's NewRelease NotesSip 1 1 5 MlDocumentationSIP Server Deployment GuideLearn how to install and configure SIP Server. Last updated on Oct 1, 2019.
Note: If you can't find the most recent released features in the Deployment Guide, see this Supplement.
High-Availability Deployment GuideLearn about setting up SIP Server in HA mode.
Integration ReferenceLearn about integrating SIP Server with SIP softswitches, gateways, and SIP Endpoints.
SIP Proxy Deployment GuideLearn about deploying SIP Proxy.
Supplement to SIP Server Deployment GuideLearn about new features introduced in SIP Server 8.1.1.
SIP Server Sizing GuideDownload this tool to evaluate SIP Server and SIP Proxy CPU load and network traffic. 
SIP Cluster SolutionLearn about SIP Cluster solution architecture.
Supported SIP DevicesSp 1152Use this document for information on supported SIP devices.
Supported Operating SystemsUse this document for information on supported operating systems by SIP Server. https://bestkfiles744.weebly.com/vuescan-9-5-16-scanner-software-with-advanced-features.html.
Helpful LinksGenesys Media Server DocumentationUse these documents to deploy Genesys Media Server and enable its features.
Learn about integrating GVP 8.1, Framework 8.0, and SIP Server 7.6 and 8.0.
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Release Note

Introduced in 8.1.103.62As per GDPR requirement, on an outbound call made by a TMakeCall request through a Routing Point, SIP Server supports playing an opt-out prompt to an end user, so that the user can decide whether to opt out from recording and monitoring. This prompt can now be played in the customer's preferred language.
Introduced in 8.1.103.61When sending an ACK message with an answer SDP before terminating an agent dialog, SIP Server now includes a codecs list matching the offer SDP that is received for the dialog.
SIP Server now supports enhanced high-availability resilience for network disruptions. The following new configuration options are added: switchover-on-msml-oos and switchover-on-trunks-oos.
Introduced in 8.1.103.59Per PCI compliance, SIP Server supports sending DTMF digits to an inbound leg of the call in SIP INFO requests.
Introduced in 8.1.103.58This release improves music-on-hold handling in deployments where SIP Server is used with Secure Real-Time Transport Protocol (SRTP). Music continues playing if an agent places a call on hold. To enable this functionality, set the sip-reinvite-action configuration option to after-hold on a Trunk DN where a re-INVITE message arrives.
To troubleshoot T-Controllers (TC) communication issues, new statistics, based on periodic T-Library polls between T-Controllers, are now available. The new configuration option, tc-latency-poll-interval, is added.
Introduced in 8.1.103.56SIP Server in SIP Cluster mode now enforces authenticated access from SIP devices. To enable this functionality, set the sip-enable-strict-auth option to true on the SIP Cluster Node VOIP Service DN. This option can also be set at the SIP Server Application or at DNs of type Extension.
Introduced in 8.1.103.54SIP Server now supports an enhanced high-availability procedure where it reports a value of 0 for 'Readiness for Switchover' in an overloaded condition (CPU usage is above the threshold).
SIP Server now supports enabling the Preview Interactions feature using the TRouteCall request containing the preview-interaction key in AttributeExtensions.
Introduced in 8.1.103.53SIP Server in SIP Cluster mode: A new Active Out-of-Service (OOS) check procedure for a VoIP service or Trunk DN ensures that this DN is placed out of service only when SIP Proxy processes the SIP traffic. The following new options are added: sipp-oos-recheck and switchover-on-sipp-oos.
Introduced in 8.1.103.52SIP Server in SIP Cluster mode adds support for remote agents to use external numbers that are not provisioned in the Configuration Database. A new configuration option, replace-agent-phone, is introduced.
SIP Server can now detect a T.38 fax SDP and drop the call when using TMakePredicitveCall requests with Call Progress Detection. For SIP Server to drop the call when T.38 fax is detected, set the fax-detected option to drop.
Introduced in 8.1.103.46In a scenario where an agent places a call on hold and SIP Server receives a hold INVITE message from a Trunk (containing the 'a=inactive' SDP), SIP Server can delay sending a re-INVITE message when connecting a Music-on-Hold (MOH) service.
Introduced in 8.1.103.45In SIP Cluster deployments, because of network issues, the connection between SIP Cluster nodes can be lost and promptly restored. T-Controller, the T-Library interface, maintains a subset of agent/DN states and interconnection to other T-Controllers in the cluster. A new configuration option, tc-reconnect-timeout, handles network outages gracefully in your environment.
In SIP Cluster deployments, where SIP-based DN ownership can be moved from one data center to another and, as a result, agent states are automatically changed to the logout state, SIP Server can now restore agent states immediately. A new configuration option, agent-state-auto-restore, handles agent states gracefully in your environment.
Introduced in 8.1.103.41Support for Polycom VVX phones (VVX 501, VVX 601), including video call support for Polycom EagleEye Mini Camera (5.9.1 and later). This support requires Media Control Platform (MCP) version 9.0.019.68 or later.
Enhanced support for integration with Unify OpenScape Voice using an alternative HA deployment. See SIP Server and OpenScape Voice Integration Overview for details.
Introduced in 8.1.103.40The following new configuration options are added (see the RN for details):sip-call-id-suffix
override-domain-ruri
sip-contact-user
sip-add-via
Introduced in 8.1.103.37For Windows and Linux operating systems, SIP Server now uses its own implementation to retrieve CPU and memory usage of SIP Server processes and threads. Previously, SIP Server used LCA to retrieve CPU and memory usage.
In SIP Cluster deployments, TEvents distribution by Session Controller to the Interaction Proxy (I-Proxy) running in HA mode is optimized.
Introduced in 8.1.103.35For Genesys Engage cloud deployments only: SIP Server can treat 1pcc calls from external callers (agents behind SIP trunks) as inbound calls. For enabling this feature, contact Product Management or your Genesys representative.
The external-contact option can now be changed dynamically. Previously, restart of SIP Server was required for the changes to take effect.
SIP Server supports Oracle Linux version 7.
For SIP Cluster mode, the default value of common/enable-async-dns is changed from 0 to 1.
When running in SIP Cluster mode, SIP Server now communicates with the SIP Feature Server dial plan through a new Extended Services component (that is run in its own thread) instead of the Call Manager thread.
When running in SIP Cluster mode, SIP Server can now terminate DN ownership in a T-Controller internal module by processing a TPrivateService request containing AttributePrivateMsgID=8273 for AttributeThisDN.
Introduced in 8.1.103.28SIP Server can now disable recording and monitoring of outbound calls in SIP Cluster deployments.
Introduced in 8.1.103.26SIP Server can now enable call recording on the agent side in SIP Cluster deployments.
Introduced in 8.1.103.25In standalone mode, SIP Server now starts a timer after sending an OPTIONS request through a SIP Proxy to a DN.
SIP Server adds the ability to monitor statistics related to SIP Feature Server interactions.
Introduced in 8.1.103.24Support for Oracle Enterprise Session Border Controller AP 1100/3900/4600/6300/6350 version E-CZ8.1.0.
Support for AudioCodes 440HD, 430HD, 420HD, and 405 IP Phones with firmware version 2.2.16.
When running in SIP Cluster mode, SIP Server now offers the T-Library interface, called Smart Proxy, for event monitoring clients (such as Stat Server).
Introduced in 8.1.103.20When running in SIP Cluster mode and when After Call Work (ACW) is controlled by UserData, SIP Server now applies the ACW state to an agent correctly.
Introduced in 8.1.103.16Support for F5 Networks BIG-IP Local Traffic Manager version 12.
Introduced in 8.1.103.10Support for Avaya Session Manager version 7.x.
Introduced in 8.1.103.08Support for secure SIP signaling, or sips, in accordance with RFC 5630.
Support for CentOS Linux 7.
Introduced in 8.1.103.06Support for a TLib-based Instant Messaging session initiated by an agent with the nailed-up connection.
Introduced in 8.1.103.02When MCP sends updated user data with an empty value in the MSML Info body, SIP Server now updates the user data with the empty value in T-Library events.
Support for a new Application-level configuration option sip-elin-timeout, for the integration with the RedSky emergency call provider.
Introduced in 8.1.102.99Support for CounterPath Bria v5.1.0, including audio and video.
Introduced in 8.1.102.98Support for AudioCodes 420HD IP Phone with firmware version 2.2.12.
Introduced in 8.1.102.95Support for a new Application-level configuration option report-error-on-routing-end, to generate EventError with ErrorCode 453 and the Call has been disconnected error message.
Synchronization of multisite (ISCC) calls that start before the HA connection between the primary and backup SIP Servers is established.
Support for Windows Server 2016 operating system. See the SIP Server page in the Genesys Supported Operating Environment Reference Guide for more detailed information and a list of all supported operating systems.
Introduced in 8.1.102.93Support for remote agents and agents with nailed-up connections to use external numbers that are not provisioned in the Configuration Database.
Support for AudioCodes 405HD and 440HD IP Phone with firmware version 2.2.12.
Introduced in 8.1.102.92Enhanced support to SIP Feature Server for assigning calling profiles (such as dial plans) to Routing Point and Trunk Group DNs for predictive calls made on behalf of a Routing Point or Trunk Group DN. This functionality is required for consistency with the internal SIP Server dial plan. A new Application-level configuration option, enable-outbound-ext-dial-plan, supports this feature. This feature depends on support from a specific version of SIP Feature Server. Consult SIP Feature Server documentation for the availability of this new feature.
Introduced in 8.1.102.89Support for providing a ringback tone to 1pcc calls. This behavior applies to inbound calls from external DNs.
Introduced in 8.1.102.81Support for the SIP Event Package for Registrations, in accordance with RFC 3680.
Support for a secure connection to Message Server using the Transport Layer Security protocol.
Introduced in 8.1.102.79Support for a new Application-level option unknown-gateway-reject-code to define which SIP error code SIP Server returns when an incoming INVITE message cannot be associated with an internal device or trunk, when the enable-unknown-gateway option is set to false.
Introduced in 8.1.102.73Support for masking of DTMF information in some attributes of T-Library messages of SIP Server log files.
Support for enhanced reporting of the monitoring mode in EventUserEvent messages to ICON that includes LCT parties information.
Sip 1 1 5 EqualsFor the features introduced in previous releases, see SIP Server 8.1.x CD Readme.
Genesys SIP Server is the Genesys software component that provides an interface between your telephony hardware and the rest of the Genesys software components in your enterprise. It translates and keeps track of events and requests that come from, and are sent to the telephony device. SIP Server is a TCP/IP-based server that can also act as a messaging interface between SIP Server clients. It is the critical point in allowing your Genesys solution to facilitate and track the contacts that flow through your enterprise. 
Fonepaw android data recovery crack. Genesys SIP Server is included in various packages of the Inbound, Outbound, and Self-services solutions for Customer Engagement and the Platform and Integration solutions for Business Optimization. Contact your Genesys representative for more information.
What's NewRelease NotesSip 1 1 5 MlDocumentationSIP Server Deployment GuideLearn how to install and configure SIP Server. Last updated on Oct 1, 2019.
Note: If you can't find the most recent released features in the Deployment Guide, see this Supplement.
High-Availability Deployment GuideLearn about setting up SIP Server in HA mode.
Integration ReferenceLearn about integrating SIP Server with SIP softswitches, gateways, and SIP Endpoints.
SIP Proxy Deployment GuideLearn about deploying SIP Proxy.
Supplement to SIP Server Deployment GuideLearn about new features introduced in SIP Server 8.1.1.
SIP Server Sizing GuideDownload this tool to evaluate SIP Server and SIP Proxy CPU load and network traffic. 
SIP Cluster SolutionLearn about SIP Cluster solution architecture.
Supported SIP DevicesSp 1152Use this document for information on supported SIP devices.
Supported Operating SystemsUse this document for information on supported operating systems by SIP Server. https://bestkfiles744.weebly.com/vuescan-9-5-16-scanner-software-with-advanced-features.html.
Helpful LinksGenesys Media Server DocumentationUse these documents to deploy Genesys Media Server and enable its features.
Learn about integrating GVP 8.1, Framework 8.0, and SIP Server 7.6 and 8.0.
Sp 1157
Release NoteDocumentationLearn how to install and configure SIP Server.
Learn about setting up SIP Server in HA mode.
Learn about integrating SIP Server with SIP softswitches and gateways.
Other documentation where SIP Server is usedLearn how to install and configure the Proactive Contact solution.
Learn about integrating GVP 8.0, Framework 8.0, and SIP Server 7.6.
Readme and AdvisoriesSIP Server 7.6 entered End of Life (EOL) on May 31, 2019. It enters End of Maintenance on November 30, 2020, and reaches End of Support on May 31, 2021. If you have questions, contact your account representative.
Release NoteDocumentationLearn how to install and configure SIP Server.
Learn about integrating SIP Server with SIP softswitches and gateways.
Learn about the Instant Messaging solution.
Readme and AdvisoriesRelease NotesDocumentationLearn how to install and configure Network SIP Server.
Learn about Genesys 7.5 and SIP Server integration.

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