GenModule102

Communication Server

Description

Communication Server: The core telephony process.

Release notes

Show release notes

Version 3.15.4 - Deprecated

Bugfix: Avoid safetynet false positive (M17031)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.15.3 - Deprecated

Bugfix: Define multiple range of CICS (M17043)
Bugfix: Fix crash during func_odbc reload (backport) (M16670)
Bugfix: Crash when handling codec's payloads (backport) (M16969)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.15.2 - Deprecated

Bugfix: Conf bridge announcements not played in the user's language (backport) (M16253)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.15.1 - Deprecated

Bugfix: Process crashed in case of ccOriginate API call (M16768)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.15.0 - Deprecated

Feature: Call swapping and transfer on media-strict PBX (M16247)
Improvement: Packaging compatible with APT (M16327)
Improvement: Ability to not send ISUP FAR message (M16226)
Improvement: HTTP action func_curl now allows multiple headers (M16333)
Improvement: Support recent cipher suites for tls (M15787)
Bugfix: Memory leak when error on outgoing MWI suscription (M16288)
Bugfix: Leak and global lock in HTTP action func_curl removed (M16333)
Bugfix: Parsing crash of diversion header (M16349)
Deprecated: Process crashes in case of ccOriginate API call (M16768)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.14.3 - Deprecated

Bugfix: Custom unavailable minivm greeting was never played (backport) (M16855)
Bugfix: The recorded name was not used when the option skip fallback was used (backport) (M16855)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.14.2 - Deprecated

Improvement: removed dependency with shell 1.23 (M0)
Bugfix: Process crashed in case of ccOriginate API call (M16768)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.14.1 - Deprecated

Improvement: Compatibilty with updated fail2ban (M15083)
Bugfix: Local networks are now whitelisted in fail2ban (M16059)
Bugfix: Solve crashes when parsing JSON (M15859)
Bugfix: Avoid codec negotiation loop in Alcatel call swapping (M15862)
Bugfix: Dynamic payload change was not working (M16025)
Bugfix: Prevent SDP-version-ignore in case of codec re-negotiation (M15994)
Bugfix: Installation failed if older firewall module was present (M15989)
Deprecated: Process crashes in case of ccOriginate API call (M16768)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.14.0 - Deprecated

Feature: Add option to skip the fallback in MinivmGreet application (M14606)
Feature: Convert plugins to new library model (M14774)
Improvement: Compatibility with Chrome 52 for WebRTC external calls (M15479)
Improvement: Add firewall rules (M15082)
Bugfix: Fix small memory leak on every WebRTC call (M15479)
Bugfix: Minivm tmp folder was not always created (M13783)
Bugfix: Fix to get the sub directory (tmp) in minivm (M13783)
Bugfix: Crash occurring when parsing the SIP message 181 forwarding (M15369)
Deprecated: Higher version provide significant improvements (M0)
Potential update impact level 1 : no critical impact expected. Update can be applied without risk of breaking critical functionality.: You will have to upgrade the Shell module and other modules too. Please verify the release notes of the Shell module 1.23.0 to find out which modules need to be upgraded. (M14774)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Shell Module >= 1.23.0
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 3.0.0 (optional: if used)

Version 3.13.0 - Deprecated

Feature: MiniVM Support (M13781)
Feature: MiniVM folder management support (M13780)
Feature: MiniVM greeting management support (M13671)
Feature: Fail2ban support (M14743)
Feature: Support of custom CDR fields on ODBC (M13662)
Feature: Early Media Suppression (M13756)
Bugfix: Reload was needed to apply linear queue strategy (M13530)
Bugfix: Fix WebRTC hold regression (M14070)
Bugfix: When remote end sends a reinvite to change the codec, transcoding was not reinitialized (M13624)
Bugfix: When remote end answers to a reinvite with several codecs, the channel codec was not the first codec proposed in the SDP (M13624)
Bugfix: Process coredumps were not created in some cases (M13582)
Bugfix: Misconfigured presence subscriptions could leak memory (M14357)
Bugfix: RTP port not freed after fast ICE failure (M14502)
Bugfix: NAT detection not always working (M14090)
Bugfix: It was not possible to force the Caller ID in the Originate when the connected line is updated (M14727)
Bugfix: Process failed to start after a fresh installation (M14951)
Potential update impact level 1 : no critical impact expected. Update can be applied without risk of breaking critical functionality.: In-band ringing setting "never" is now stricter. Must be changed to "no" to keep previous behavior (M13756)
Deprecated: Higher version provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- System Base >= 1.10.1 (optional: only required when coredumps are necessary)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)
- Firewall Module >= 2.1.0 (optional: for fail2ban feature)

Version 3.12.0 - Deprecated

Improvement: Scalability and robustness registration flags (M11512)
Improvement: Do not register at server startup (M13791)
Bugfix: recvonly was missing in the SDP of the SIP trunk when a sendonly was received (M13299)
Bugfix: RTP probation was not working (M12797)
Bugfix: escall was not working when ulaw was not configure in the module (M13476)
Bugfix: Define the maximum number of calls (M13602)
Deprecated: Process failed to start after a fresh installation (M14951)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)
- SIP Selftest Probe >= 2.0.0 (optional: if used)

Version 3.11.10 - Early deployment

Improvement: Move ha scripts to asterisk (backport) (M17031)
Bugfix: RTP ports were not freed after connectivity loss while ringing (backport) (M17516)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.9 - General deployment

Bugfix: Fix crash during func_odbc reload (backport) (M16670)
Improvement: Allow to configure HTTP bind address (backport) (M16508)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.8 - Deprecated

Bugfix: T38 was only working on eth0 (M16616)
Deprecated: Apply changes fails because of syntax error in perl_code
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.7 - Deprecated

Bugfix: Crash parsing diversion header (M16349)
Deprecated: Apply changes fails because of syntax error in perl_code
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.6 - Early deployment

Bugfix: Solve crashes when parsing JSON (backport) (M15859)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.5 - Early deployment

Improvement: Compatibility with Chrome 52 for WebRTC external calls (backport) (M15479)
Bugfix: Fix small memory leak on every WebRTC call (backport) (M15479)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.4 - Early deployment

Bugfix: RTP port not freed after fast ICE failure (backport) (M14502)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.3 - Early deployment

Improvement: WebSocket connections more stable (backport) (M14354)
Bugfix: Misconfigured presence subscriptions could leak memory (backport) (M14357)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.2 - Early deployment

Bugfix: Reload was needed to apply linear queue strategy (backport) (M13530)
Bugfix: Update ICE stack fixing some crashes (backport) (M14190)
Bugfix: Fix WebRTC hold regression (backport) (M14070)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.1 - Early deployment

Bugfix: escall was not working when ulaw was not configure in the module (backport) (M13476)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.11.0 - Early deployment

Feature: Support REFER on own callid (M12936)
Feature: New application SIPMute (M12935)
Improvement: Never expose internal DNS aliases in outgoing requests (M13244)
Improvement: Refresh MWI subscription on DNS update (M13374)
Bugfix: RTP port exhaustion (M13145)
Bugfix: MWI was using DNS even when not needed (M13180)
Bugfix: Voice grammar in Dutch was incorrect (M13377)
Bugfix: Externnotify was always activated (M13401)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.10.1
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4.2
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.10.0 - Early deployment

Feature: DTMF based transfer can now send REFER on original call (M12492)
Bugfix: Syntax to say time was incorrect in French and in Spanish (M13054)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.9.0 - Early deployment

Feature: adding response to EC ping on websocket (M12824)
Improvement: Use better From header in Endpoint Abstraction register (M12859)
Bugfix: Always keep retrying MWI subscription (M12491)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.8.0 - Early deployment

Feature: Custom internal DNS entry (M12728)
Improvement: Threadpool options are now configurable (M11201)
Improvement: Support incoming SIP REFER from Lync (M12617)
Improvement: Re-register on DNS update (M12745)
Bugfix: Realtime peers using WebSocket were unreachable after reload (M12574)
Bugfix: Make notification on VoicemailMenu deactivable (M12601)
Bugfix: Not able to record voicemail (M12800)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4
- VMXML Module >= 2.0.0 (optional: if used)

Version 3.7.0 - Early deployment

Feature: Multi-threaded task processors for messaging and presence (M11195)
Feature: Make RTP binding address configurable, per peer (M12369)
Feature: Limit size of message queue (M11201)
Feature: New HINT_WATCHERS function (M11904)
Improvement: Use correct encoding for instant messaging (M12334)
Improvement: React on first watcher of a hint (M12395)
Bugfix: Base64-encoded presence data was truncated (M12398)(M12526)
Bugfix: FMU sync process could fail to run completely (M12627)
Bugfix: Restore the old pincode prompt when joining a conference call (M11684)
Bugfix: ICE could fail because of too many candidates
Bugfix: Try to avoid hanging WebSocket connections
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.6.2 - Early deployment

Bugfix: MWI was using DNS even when not needed (backport) (M13180)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.6.1 - Early deployment

Feature: DTMF based transfer can now send REFER on original call (backport) (M12492)
Bugfix: Always keep retrying MWI subscription (backport) (M12491)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.6.0 - Early deployment

Feature: Add Google Cloud Messaging support to XMPP module (M11450)
Feature: Subscribe to Message Waiting Indicator in EPA (M11156)
Feature: Support dynamic EPA registration (M11071)
Feature: New JSON presence format (M11901)
Improvement: Remove realtime priority by default, reducing load peaks (M11182)
Improvement: Added user=phone in From PAI, RPID and Diversion headers (M11422)
Improvement: Make http session limit configurable (M11175)
Improvement: Parse contact URI arguments (M10528)
Improvement: All the allowed codecs are now proposed in the reinvite so that the remote peer should always accept the SDP (M11604)
Improvement: Expose mailbox information in MWI notify callflow (M11969)
Improvement: Support out-of-order notify matching an outstanding subscription (M11822)
Improvement: Subscribe action offers more options and reports status (M12121)
Improvement: Prevent remote bridge when both end does not negotiate the same capabilities (M11793)
Bugfix: Received REINVITE with new codec were not propagated (M10416)
Bugfix: Blind transfers were not logged properly in queue_log (M10886)
Bugfix: PAI should only provide SIP URI format in output (M11245)
Bugfix: Avoid duplicating MWI subscriptions on reload (M11156)
Bugfix: Domain in the PAI for COLP were incorrect (M11245)
Bugfix: Authorization header could be corrupted (M11005)
Bugfix: Upstream fixes for DTLS and messages (M11335)
Bugfix: Diversion reason lost (M11245)
Limitation: iLBC support has been deprecated. G.729 is recommended for low bandwidth environments
Bugfix: Support dynamic nature of address whatever the ss7 prefix configuration (M11804)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.5.2 - Early deployment

Bugfix: Make RTP binding address configurable (backport) (M12369)
Bugfix: Update DTLS crypto to fix incoming calls in Firefox 39+ (M11873)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.5.1 - Early deployment

Improvement: Make http session limit configurable (M11175)
Bugfix: Upstream fixes for DTLS and messages (M11335)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.5.0 - Early deployment

Feature: New upstream version (M9933)
Feature: Support registering on multiple PBX using Abstracted Endpoint 2 (M10273)
Feature: Added the voicemail volume gain parameter (M9914)
Feature: Add xmpp configuration (M10373)
Improvement: Add privacy in diversion header (M10216)
Improvement: Added the possibility to only trust the PAI header (M10158)
Improvement: Specific certificates for DTLS-SRTP (M9931)
Improvement: Added request URI when pushing NOTIFY to the callflow (M10458)
Improvement: Log microseconds (M10443)
Improvement: Update ICE stack (M10936)
Bugfix: Upgrade libssl (M10052)
Bugfix: It was not possible to transfer video call using dynamic payload such as H264 (M8954)
Bugfix: Don't start asterisk when sipsocket fails to bind (M9934)
Bugfix: Video not working when dual ring of phone with video support and another one without (M9919)
Bugfix: No new WebSocket connection would sometimes be accepted (M9877)
Bugfix: Process could crash when detecting fax tone (M10100)
Bugfix: Recurrent crash during DTLS handshake (M10104)
Bugfix: IAXVARS could make asterisk crash (M10064)
Bugfix: When receiving an 480 SIP message the dialstatus variable is in congestion (M10443)
Bugfix: Audio issue when REINVITE with specific codec is rejected by remote (M10416)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.4

Version 3.4.7 - Deprecated

Bugfix: Memory leak when error on outgoing MWI subscription (backport) (M16288)
Deprecated: Asterisk could crash in case of an attended transfer (M20164)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.6 - Deprecated

Bugfix: Encode uri of "refer-to" field during the transfer (M15563)
Bugfix: Set the connected line and the peer IP in "refer-to" field during the transfer (M15563)
Deprecated: Asterisk could crash in case of an attended transfer (M20164)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.5 - Deprecated

Feature: Early Media Suppression (backport) (M13756)
Deprecated: Asterisk could crash in case of an attended transfer (M20164)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.4 - Deprecated

Improvement: Never expose internal DNS aliases in outgoing registrations (backport) (M13244)
Bugfix: The SIPTRANSFER_REFERER variable did not contain the complete refer-by header (backport) (M13750)
Deprecated: Asterisk could crash in case of an attended transfer (M20164)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.3 - Early deployment

Feature: Subscribe to Message Waiting Indicator in EPA (M12076 Backport) (M11156)
Feature: Custom internal DNS entry (M12728 Backport)
Improvement: Reregister on DNS update (M12745 Backport)
Improvement: Support out-of-order notify matching an outstanding subscription (M12076 Backport) (M11822)
Improvement: Support incoming SIP REFER from Lync (M12617 Backport)
Bugfix: FMU synchronization mechanism was not always working (M12627 Backport)
Bugfix: Crash during the notify contact parsing (M12939)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.2 - General deployment

Improvement: Added user=phone in From PAI, RPID and Diversion headers (Backport M11422)
Bugfix: Incorrect domain in COLP PAI (Backport M11245)
Bugfix: PAI should only provide sip uri format in output (Backport M11245)
Bugfix: Diversion reason lost (Backport M11245)
Bugfix: Avoid duplicating MWI subscriptions on reload (Backport M11156)
Dependency:
- Baseline >= 2.x.x and >= 3.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.1 - General deployment

Feature: Support registering on multiple PBX using Abstracted Endpoint 2 (Backport M10273) (M10488)
Improvement: Added the possibility to only trust the PAI header (Backport M10158)
Improvement: Add privacy in diversion header(M10216)
Improvement: Add the possibility to define maximum number of calls (M10477)
Bugfix: Authorization header could be corrupted (M11005)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.4.0 - Early deployment

Improvement: Added the possibility to configure the outbound registration timeout (M9699)
Improvement: Added the ability to modify SIP retransmission timers (M9720)
Bugfix: When using the custom transfer via the '*' the called was able to trigger the transfer (M9705)
Bugfix: SIP trunks using sipsocket and qualify could stay stuck in unreachable mode (M9888)
Bugfix: When receiving a maddr address we should not resolve the sent-by host in the via header (M9806)
Bugfix: Deactivate defense daemon when switching to standby mode (M9721)
Feature: Expanded support of message content-types when outside context of a call (M9875)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.15 - Early deployment

Bugfix: Fixed destination number on TRANSFER event in queue log (M19497)
Bugfix: Blind transfers were not logged properly in queue_log (M10886)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- System Base Module >= 1.10.1 (optional: only required if coredumps are necessary)
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.14 - Early deployment

Improvement: In case of ringall ring directly member when they get available (M18271)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- System Base Module >= 1.10.1 (optional: only required if coredumps are necessary)
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.13 - Early deployment

Bugfix: REINVITE with codec rordering were not propagated (backport) (M17405)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- System Base Module >= 1.10.1 (optional: only required if coredumps are necessary)
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.12 - General deployment

Bugfix: Prevent SDP-version-ignore in case of codec re-negotiation (Backport) (M15994)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- System Base Module >= 1.10.1 (optional: only required if coredumps are necessary)
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.11 - General deployment

Feature: Support of custom CDR fields on ODBC (backport) (M13662)
Bugfix: Coredumps were not generated in some cases (M13582)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- System Base Module >= 1.10.1 (optional: only required if coredumps are necessary)
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.10 - General deployment

Bugfix: Reload was needed to apply linear queue strategy (Backport M13530)
Bugfix: When remote end sends a reinvite to change the codec, transcoding was not reinitialized (Backport M13624)
Bugfix: When remote end answers to a reinvite with several codecs, the channel codec was not the first codec proposed in the SDP (Backport M13624)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.9 - General deployment

Bugfix: recvonly was missing in the SDP of the SIP trunk when a sendonly was received (M13299)
Bugfix: When remote end answers with several codecs many reinvite was sent (M13624)
Bugfix: Maxcalls was not configurable (M13620)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.8 - General deployment

Bugfix: SIP stack blocked when remote end answers a reinvite with several codecs (Backport M13343)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.7 - Deprecated

Bugfix: Restored the old pincode prompt when joining a conference call - Backport (M11684)
Deprecated: SIP stack blocked when remote end answers a reinvite with several codecs (M13343)
Dependency:
- Baseline = 2.x.x
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.6 - Deprecated

Improvement: All the allowed codecs are now proposed in the reinvite so that the remote peer should always accept the SDP (Backport M11604)
Bugfix: Audio issue when REINVITE with specific codec is rejected by remote end (Backport M10416)
Bugfix: Received REINVITE with new codec were not propagated (Backport M10416)
Bugfix: It was not possible to transfer video call using dynamic payload such as H264 (Backport M8954)
Bugfix: Video was not working when doing a dual ringing of a phone with support and other one without (Backport M9919)
Deprecated: SIP stack blocked when remote end answers a reinvite with several codecs (M13343)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)
- Snom 300 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 320 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 360 Resource >= v3.13.1 (optional: upgrade if used)
- Snom 370 Resource >= v3.13.1 (optional: upgrade if used)

Version 3.3.5 - General deployment

Bugfix: Process could crash when the action SetRemoteVar was used on a IAX channel (Backport M10064)
Bugfix: SIP trunks using sipsocket and qualify could stay stuck in unreachable mode (Backport M9888)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.4 - General deployment

Bugfix: Don't start asterisk when sipsocket fails to bind (Backport of M9934)
Bugfix: Process could have crashed when detecting afax tone (Backport of M10100)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.3 - General deployment

Bugfix: Security upgrade of libssl for WebRTC (M9639)
Bugfix: Video not stopped when call is put on hold (M9350)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.2 - Deprecated

Improvement: Add person activity to presence notify (M9371)
Improvement: The channel variable FORWARD_CHANNEL is now set when the Dial Asterisk Options 'i' is used (M9390)
Bugfix: Leftover presence channels could stay in the channels list (M9545)
Bugfix: Video not stopped when call is put on hold (M9350)
Deprecated: Crash when putting a call on hold (M9350)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.1 - General deployment

Bugfix: Set the correct RTP source address when using SIP sockets (M9305)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.3.0 - Early deployment

Improvement: Propagate hangup cause on CANCEL messages (M9224)
Improvement: Smaller MTU for DTLS handshake (M9320)
Bugfix: Increase size of account code field in CDR to 255 characters (M9264)
Bugfix: The music on Hold was not stopped after the call transfer to a queue's agent (M7492)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.2.0 - Early deployment

Feature: Add WebRTC support (M8955)
Feature: Add DTLS-SRTP support for WebRTC in Firefox and newer Chrome (M8767)
Feature: ICE and RTP configuration to work behind NAT (M9117)
Feature: Notify m.Connect when a registration is overwritten (M8893)
Feature: New action Subscribe usable in callflows (M8676)
Feature: Notify presence events to m.Connect (M8677)
Feature: Allow voicemail configuration per extension (M8780)
Improvement: Reduce module installation time on high latency networks (M8766)
Improvement: Prevent asterisk to start without binding on requested SIP or IAX addresses (M8732)
Improvement: Prevent uncontrolled growth of log files (M8400)
Improvement: Avoid blocking indefinitely on FastAGI callflow actions (M7462)
Improvement: Latest security fixes (M9027)
Improvement: Asterisk reload was slow when using several thousands global variables (M8855)
Improvement: Unused configuration global variables were not removed upon reload (M9062)
Improvement: Added the possibility to deactivate the Jitter Buffer when bridging two IAX channels (M8083)
Improvement: Added the possibility to configure the maximum number of simultaneous calls (M9139)
Bugfix: There was a one way audio issue when dynamic feature and direct rtp setup were used Together (M8964)
Bugfix: The custom DTMF based call transfer was not woring since version 3.0 (M9084)
Bugfix: Video was wrongly offered in outgoing invite even when caller did not support it (M8050)
Bugfix: Activating the Jitter Buffer and the IAX trunking was causing audio quality issue (M8083)
Bugfix: There was a one way audio when calling a device which answers with two codecs and when DirectMedia is disabled (M9083)
Bugfix: No Directmedia when calling a phone in G729 only via a Queue and when the caller supports COLP (M9083)
Bugfix: The SDP in the OK could be different from the SDP provided in the provisional call progress (M9083)
Bugfix: The process could crash when confirming an attened transfer while the consulted destination was still ringing (M9130)
Bugfix: Fixed memory leak which occured when making call and answering them (M9202)
Bugfix: Additional SIP sockets were not compatible with dynamic address on a SIP trunk (M9187)
Bugfix: There was stale channels when using the DTMF based call transfer (M9292)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.9.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1
- Call Management Module >= 1.7.4 (option: if used)
- STARTDYNAMICAPPLICATION Action >= 8.0 (optional if Call Management Module is used)

Version 3.1.0 - Deprecated

Feature: Support for preferred devices in Queues (M5968)
Feature: Configurable email return-path header (M2601)
Feature: Support for remote party presence/extended presence (M7384)
Feature: Voicemail with less options (M8032)
Feature: Detection of voicemail outcome: minimum length and maximum silence of voicemails are now configurable (M7996)
Feature: Add video support on Communication Server 3 (M7912)
Improvement: Use escaux-minutely for minutely cronjobs (M7759)
Improvement: Register to a remote peer when using a SIP socket (M7903)
Improvement: Add a timeout for external scripts (AGI) (M7462)
Bugfix: Parsing of "Additional SIP sockets" was not working in some cases, with multiple adresses (M7903)
Bugfix: In some call transfer scenarios old mediastreams could remain in the list (M7489)
Bugfix: Disabling voicemail to e-mail was not possible
Bugfix: Hangup cause was not always propagated by the Dial application (M8129)
Bugfix: Reason header was not parsed in the Bye message (M8129)
Bugfix: Asterisk was stopping trying to register SIP Trunks after receiving a 403 (M7917)
Bugfix: An invalid host value for a peer could not be corrected by a reload (M8144)
Bugfix: The variable SIP_CODEC was not taken into account for inbound call (M8388)
Bugfix: No ringback tone when dialing a phone with a new codec after answering a call (M8257)
Bugfix: Multi-codec issue when dynamic feature or recording is used (M6921)
Bugfix: In a multi codec environment the reinvite was not always sent to set up the call in peer to peer
Bugfix: Originate failed because the macro could loose their context (M8463)
Improvement: Register to a remote peer when using a SIP socket (M7903)
Bugfix: Parsing of "Additional SIP sockets" was not working in some cases, with multiple adresses (M7903)
Bugfix: In some call transfer scenarios old mediastreams could remain in the list (M7489)
Bugfix: Disabling email to voicemail was not possible (M0)
Bugfix: The iax trunk frequency set did not always match the one configured (M8044)
Bugfix: The process could crash after a reinvite (M9024)
Limitation: When using H264 and DirectMedia together, video stream might not be working after a call transfer (M9087)
Deprecated: Higher versions provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.4.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 3.0.5 - Deprecated

Bugfix: Memory leak in advanced codec negotiation (M8552)
Deprecated: Higher versions provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.4.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 3.0.4 - Deprecated

Feature: Setting specific channel variables with SIP INFO messages (M7712)
Deprecated: Higher versions provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.4.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 3.0.3 - Deprecated

Feature: Add SIP overload control and blacklist (M7539)
Bugfix: Memory leak and locking problems in external notify (M7648)
Bugfix: Potential crash in external notify (M7479)
Bugfix: Potential crash in net.Console support due to race condition (M7731)
Bugfix: Add missing config file for T.38 support
Deprecated: Higher versions provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.4.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 3.0.2 - Deprecated

Bugfix: Put chan_dahdi in a separate package (M7640)
Deprecated: Higher versions provide significant improvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.8.0
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 3.0.0 - Deprecated

Feature: Limited support for SS7 (M6919)
Improvement: Conference without a dependency on Dahdi (M7345)
Deprecated: Higher versions provide significant imporvements (M0)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.2.0 (optional: upgrade if used)
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.7 - Early deployment

Bugfix: Video lost when SIP Session Timers is activated (M19546)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.6 - Early deployment

Improvement: Removed dependency on DAHDI module (M19147)
Potential update impact level 2 : in the event this update contains a bug, it might have critical impact. Respect dependencies and retest your most important callflows and applicative integrations.: This removes support for the AddToConference action (M19147)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.5 - Early deployment

Improvement: activation of RTP probation (Backport M12797)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0 (not applicable in a vSOP guest environment)
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.4 - Early deployment

Improvement: Added the ability to modify SIP timers (M9720)
Bugfix: Deactivate defense daemon when switching to standby mode (M9721)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0 (not applicable in a vSOP guest environment)
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.3 - General deployment

Improvement: Prevent asterisk to start without binding on request SIP or IAX addresses (M8732)
Improvement: Added the possibility to add polycom specific preemption reason header (M7301)
Improvement: Added the possibility to deactivate the Jitter Buffer when bridging two IAX channels (M8083)
Improvement: Added the possibility to configure the maximum number of simultaneous calls (M9139)
Improvement: Ignore the installation of Dahdi on a VSOP (M9078)
Bugfix: Activating the Jitter Buffer and the IAX trunking was causing audio quality issue (M8083)
Bugfix: There was a one way audio when calling a device which answers with two codecs and when DirectMedia is disabled (M9083)
Bugfix: No Directmedia when calling a phone in G729 only via a Queue and when the caller supports COLP (M9083)
Bugfix: The SDP in the OK could be different from the SDP provided in the provisional call progress (M9083)
Bugfix: Additional SIP sockets were not compatible with dynamic address on a SIP trunk (M9187)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0 (not applicable in a vSOP guest environment)
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.2 - Early deployment

Feature: Relax dialog id check in SIP dialogues (M8381)
Improvement: Reduce module installation time on high latency networks (M8766)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.1 - General deployment

Improvement: Add a timeout for external scripts (AGI) (M7462)
Bugfix: An invalid host value for a peer could not be corrected by a reload (M8144)
Bugfix: Minmessage and maxsilence parameters for voicemail were sometimes not correctly set (M7996)
Bugfix: Incorrect hangup direction in Hangup manager event (M7782)
Bugfix: No ringback tone when dialing a phone with a new codec after answering a call (M8257)
Bugfix: Multi-codec issue when dynamic feature or recording is used (M6921)
Bugfix: In a multi codec environment the reinvite was not always sent to set up the call in peer to peer
Bugfix: The variable SIP_CODEC was not taken into account for inbound call (M8388)
Bugfix: Originate failed because the macro could loose their context (M8463)
Bugfix: Call transfer using '*' unstable (M8209)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.8.0 - General deployment

Feature: Be able to configure Return Path for email sent by asterisk (M2601)
Feature: Added support for remote party presence/extended presence (M7384)
Feature: Detection of voicemail outcome: minimum length and maximum silence of voicemails are now configurable (M7996)
Improvement: Use escaux-minutely for minutely cronjobs (M7759)
Bugfix: Parsing of "Additional SIP sockets" was not working in some cases, with multiple addresses (M7903)
Bugfix: In some call transfer scenarios old mediastreams could remain in the list (M7489)
Bugfix: Hangup cause was not always propagated by the Dial application (M8129)
Bugfix: Reason header was not parsed in the Bye message (M8129)
Bugfix: Asterisk was stopping trying to register SIP Trunks after receiving a 403 (M7917)
Bugfix: The iax trunk frequency set did not always match the one configured (M8044)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.7.0 - Deprecated

Bugfix: Dynamic rtp payload type were not negotiated properly when directmedia is used (M7696)
Bugfix: H263+ not working because of a typo in the SDP (M7906)
Bugfix: incoherent codec negotiation in case of video support in a multi-audio code environment (M7911)
Bugfix: When using numeric DSCP, the TOS is computed with the TOS=4*DSCP and the assumption that ECN=0 (M7784)
Bugfix: DSCP was not set on Video RTP (M7784)
Feature: Add voicemail lite and voicemail minimal to be used with VoicemailMenu action. (M7148)
Deprecated: Voicemail for new users is not working until the process is manually reloaded or restarted
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.6.1 - General deployment

Bugfix: Video was not working when using the variable SIP_CODEC (M7518)
Bugfix: The caller video codec was not preserved when transiting on sip trunks (M7518)
Bugfix: Memory leak and locking problems in external notify (M7648)
Bugfix: Potential crash in external notify (M7479)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.6.0 - General deployment

Feature: optional access to the Manager Interface (M7341)
Bugfix: There was no ringback tone or music on hold in case of codec change after an answer or a call progress (M7326)
Bugfix: channelGroup not cleanup after a call transfer via the API (M7418)
Bugfix: The buffer refer_to was too small (M7401)
Bugfix: There was no ring tone after a blind transfer via net.Console or net.Desktop (M7488)
Bugfix: The deadlock prevention script was not started automatically (M7509)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.5.0 - General deployment

Feature: Configurable register timers for IAX (M7069)
Feature: Configurable register timers for SIP (M7070)
Feature: Added the possibility to configure the IAX trunk frequency (M6951)
Feature: Allow decimal values for Differentiated Services fields (M7071)
Improvement: Relax DTMF is now always set to yes
Improvement: Default SIP Timer T1 for the re-transmission is now a fix value of 500 ms (M6777)
Improvement: Removed hardcoded callflows for extensions 8500,8501,8502,8505 (M6701)
Limitation: Core dump analysis is not possible in this version due to missing debug symbols
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.6 - General deployment

Bugfix: Originate failed because the macro could loose their context (M8463)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.5 - General deployment

Bugfix: Incorrect hangup direction in Hangup manager event (M7782)
Bugfix: No ringback tone when dialing a phone with a new codec after answering a call (M8257)
Bugfix: Multi-codec issue when dynamic feature or recording is used (M6921)
Bugfix: In a multi codec environment the reinvite was not always sent to set up the call in peer to peer
Bugfix: The variable SIP_CODEC was not taken into account for inbound call (M8388)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.4 - General deployment

Bugfix: There was no ringback tone or music on hold in case of codec change after an answer or a call progress (M7326)
Bugfix: There was no ring tone after a blind transfer via net.Console or net.Desktop (M7488)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.3 - General deployment

Bugfix: Some channels were never removed from the ChannelGroup (M7418)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.2 - General deployment

Bugfix: Fix potential deadlock on call transfer (M6636)
Bugfix: Inband ringback tone not decoded by some end-point after a dynamic codec change (M7149)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.1 - General deployment

Bugfix: Added missing G.729 SOP Shell plugin (M0006242)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.4.0 - Deprecated

Feature: Support of Direct RTP Setup. Backward compatible with SIP canreinvite parameter but set SIP Direct Media to 're-invite' in order to preserve the configuration (M6628)
Feature: SIP trunk on other sockets than the default one (M6625)
Feature: Call Admission Control check can be done in a callflow (M6726)
Feature: Call Admission Control now support IAX as well (M6734)
Improvement: Call Admission Control is a separate module (M6750)
Improvement: Export media streams events through AMI (M6807)
Improvement: Several SIP trunks on the same socket (M6694)
Improvement: Support initiating transfers from the NAG to a CS2 PBX (M6636)
Improvement: Add relax dtmf option (M6637)
Bugfix: In case of successive REINVITE with codec change, the OK answer was not correct (M6640)
Bugfix: The transcoding was not triggered in case of REINVITE with codec change to a codec not support by the remote peer (M6206)
Deprecated: Potential deadlock on call transfer (M6636)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1

Version 2.3.8 - General deployment

Bugfix: Locking problem in external notify (M7648)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1 (optional: if TLS is used)

Version 2.3.6 - Early deployment

Bugfix: Fix potential deadlock on call transfer (M6636)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1 (optional: if TLS is used)

Version 2.3.5 - Deprecated

Bugfix: the call transfer via DTMF failed from time to time (M7065)
Deprecated: Potential deadlock on call transfer introduced by M6636 (M6636)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1 (optional: if TLS is used)

Version 2.3.4 - Deprecated

Improvement: Clearer medialinks usage output (M6269)
Bugfix: Some fax machines were unable to transmit/receive when the T.38 gateway was active due to and outdated dsp library (M6746)
Bugfix: Intra-site calls were counted as inter-site calls when not using reinvite (M6268)
Bugfix: SIP selftest was incrementing medialink usage (M6270)
Bugfix: Parameter for the maximum call duration was not properly used (M6431)
Bugfix: fixed codecs negociation issue for video calls (M6437)
Bugfix: SIP Refer messages sent by NAG were not RFC3261 compliant (M6636)
Deprecated: Potential deadlock on call transfer introduced by M6636 (M6636)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1 (optional: if TLS is used)

Version 2.3.3 - General deployment

Improvement: Added the possibility to enable TLS (M6261)
Improvement: Added the possiblity to send notify message to Aastra phones (M6087)
Improvement: Productize the extern notify script (M6345)
Improvement: The preferred codec is now the only codec proposed in the 200 OK and in the RE-INVITE (M6251)
Improvement: All compatible codecs are now offered in the outgoing re-invite in order not to be too restrictive on a SIP trunk (M6251)
Improvement: If a re-invite is rejected with a 488 Not Acceptable, a fallback to bridge in asterisk is done automatically (M6251)
Bugfix: No connected line update for the initial caller after API transfer (M6307)
Bugfix: Native channel codec was changed even if reinvite was disabled (M6251)
Bugfix: There was a segmentation fault when using asterisk command line after upgrading if restart was not done (M6430)
Bugfix: Music on hold with mp3 files was not working when mode "Play the music from the beginning" was selected in the music on hold resource (M6507)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0
- Web Server >= 1.1.1 (optional: if TLS is used)

Version 2.3.2 - General deployment

Improvement: G729 requirements are now prepackaged in the module (M6241)
Bugfix: The Call Admission Control option was not taken into account (M6252)
Bugfix: G729 shell plugin was not working (M6242)
Bugfix: The call could be cut after a API call to redirect both channel in a bridge (M6175)
Bugfix: Apply callflow changes was not working (M6110)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0

Version 2.3.1 - General deployment

Bugfix: Fixed potential crash when accepting a call from a queue (M6234)
Limitation: "Enable Call Admission Control" option doesn't work (M6252)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0

Version 2.3.0 - Deprecated

Feature: Added Midcallflow triggering (M5882)
Feature: Added defence mechanism in order to restart asterisk in case of deadlock in the SIP stack (M5888)
Feature: Ported legacy asterisk 1.2 CDR in order to preserve SMP advanced reporting and net.Desktop CallHistory (M5972)
Feature: Added multi-codec support in case of re-invite (M5868)
Feature: Added advanced codec negotiation in order to reduce the need of transcoding (M5868)
Feature: Enable REFER on third party asterisk (M5582)
Bugfix: After a ForkCDR, it was not possible to change/set CDR variables (M5881)
Bugfix: ForkCDR was not written to the cdr table (M6039)
Bugfix: The first prompt sound in some action such as the IVR was always in english (M0006028)
Bugfix: Added missing templates for notify-exten and notify-context debconf (M0)
Bugfix: The music on-hold on a SIP trunk was sometimes inconsistent(M5960)
Bugfix: The caller ID after a blind transfer was sometimes incorrect (M5939)
Deprecated: Process can crash when a call sent by a queue is answered (M6234)
Deprecated: "Enable Call Admission Control" option doesn't work (M6252)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.7
- Sounds Module >= 1.8.0
- SOP API Module >= 4.1.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)
- Watchdog Module >= 1.1.0

Version 2.1.2 - General deployment

Bugfix: Voicemail maximum message length was incorrectly interpreted in seconds instead of minutes. (M0005886)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.2
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.3
- Sounds Module >= 1.8.0
- SOP API Module >= 4.0.2 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)

Version 2.1.1 - Early deployment

Bugfix: Filesystem permissions on some files were not correct when upgrading from certain versions of the Asterisk-1.2x module. (M5829)
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.2
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.3
- Sounds Module >= 1.8.0
- SOP API Module >= 4.0.2 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)

Version 2.1.0 - General deployment

Bugfix: Callcounters on SIP channels were disabled which caused CheckDeviceAvailability to always detect a device as being idle.
Bugfix: Process crashed when using the console dial command.
Bugfix: Process crashed when doing a blind transfer with net.Console.
Feature: Added a plugin to restart Communication Server from the shell.
Improvement: Disabled some unused modules.
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.1
- SOP Base Module >= 1.4.3
- Sounds Module >= 1.8.0
- SOP API Module >= 4.0.2 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)

Version 2.0.0 - Deprecated

Feature: Potential update impact level 3 : in the event this update contains a bug, it might have critical impact. Given the complexity of the update, it is advised to contact ESCAUX support before applying this update. initial release. Successor for Asterisk-1.2 Module.
Deprecated: Internal, non-public release.
Dependency:
- Baseline >= 2.0.0
- Database Server Module >= 2.2.0
- Dahdi Module >= 1.0.0
- SMP >= 4.8.0
- SOP Base Module >= 1.4.2
- Sounds Module >= 1.8.0
- SOP API Module >= 4.0.0 (optional: upgrade if used)
- PUM Module >= 3.2.3 (optional: upgrade if used)

Module configuration interface

create_resource_form: .:/usr/share/escaux/glue/lib:/usr/share/escaux/glue/bin/gen_wiki_documentation/src/lib:/usr/share/escaux/glue/bin/gen_wiki_documentation/src/lib/

SIP Direct Media
DNS Lookup
SIP videosupport
Log Level
Indications
Default Context (blank=default)
IAX Jitter Buffer
Qualify SIP devices registration
SIP register timers (minimum,maximum,default,timeout) in seconds
Enable voicemail to email
Voicemail email sender
WAV file voicemail email compression
Server side DSCP marking: Enter comma separated list of 4 DSCP values for SIP, RTP audio, RTP video, IAX (Eg: CS3,EF,AF41,EF)
RTP Port Range (start-end)
ICE external address
Number of park slots (default 20)
Parktime in seconds (default 45)
In-band ringing
*Group-pickup extension (default: 8)**
Maximum call duration in seconds (default: one day)
Asterisk console timeout (default: one day)
Blind transfer key
Voicemail to email click-to-dial
Voicemail to email click-to-dial hear message service (default: 8500)
Voicemail to email click-to-dial call a phone service (default: user's extension)
Sync Extension Policy
Disable IAX on external interfaces (default: no)
First codec to use (default: alaw)
Second codec to use (default: ulaw)
Third codec to use (default: none)
Fourth codec to use (default: none)
Voicemail maximum length message (in minute)
Voicemail notification (by email) language
Always fake user rejection (Make sip username more difficult to guess)
Allow guest calls
Request URI filter
Enable TCP
Enable Call Admission Control
MidCallflow triggering activation key sequence
Enable TLS
Use custom script for new voicemail
Extension assigned to the SMSService.Service callflow
Phone_id to use in the externnotify script
Voicemail extension (used for the externnotify script)
Additional SIP sockets
Use external (callflow) check for Call Admission Control
Debconf
IAX Trunk frequency (ms)
IAX register timers (minimum,maximum) in seconds
Timers definition (t1min,timert1,timerb)
Enable enriched Remote Party Presence Information
Customer manager interface password
Voicemail minimum length (in seconds)
Voicemail maximum silence (in seconds)
SIP Overload Control
Voicemail volume gain
Enable realtime
Max calls
Email Return Path

Module configuration parameters

SIP Direct Media: (was "SIP canreinvite" before v2.3.4)
- no: Media streams will always go through Communication Server. They will never be setup directly between clients.
- re-invite: When initiating a call in SIP, the INVITE message contains information on where to send the media streams. In the first INVITE, Communication Server is set as the intermediary point for media streams between clients. Once the call has been accepted, Communication Server will send another INVITE message (hence the name re-INVITE) to the clients with the necessary information for the two clients to send media streams directly to each other.
  - If one of the clients is configured with canreinvite=NO, Communication Server will not issue a re-invite at all.
  - If clients use different codecs, Communication Server will not issue a re-invite.
- direct-rtp-setup: When initiating a call, INVITE messages will already contain the necessary information for the two clients to send media streams directly to each other.
  - If one of the clients is configured with canreinvite=NO, Communication Server will not setup direct media streams.
  - If clients use different codecs, Communication Server Communication Server will not setup direct media streams.

DNS Lookup: Allow Communication Server DNS entries manager
SIP videosupport: Configure Communication Server to support video
Log Level:
Indications: tone localization.
Default Context: default context for SIP devices (blank=default allow unauthenticated SIP devices to call internally only)
IAX Jitter Buffer: enable or disable IAX jitter buffer. Enabling jitter buffer is useful when you have quality issues with IAX trunks over internet
- no: Do not activate the Jitter Buffer
- yes: Activate the jitter buffer if the bridged channel's technology does not support Jitter Buffer (SIP and IAX peers are supposed to support Jitter Buffer)
- force: Always activate the Jitter Buffer even if the bridged channel's technology supports Jitter Buffer
- force-nonativebridge: Force the Jitter Buffer only if we do not bridge two IAX channels
Qualify SIP devices registration: Default qualify configuration for SIP devices. If you turn on this option, the process will send SIP OPTIONS messages every 60s. If the device does not respond within the defined delay, it will be considered as off-line. Selecting 'no' will disable this functionality. Selecting 'yes' will use a good default timeout value (2000 ms). A custom timeout delay (expressed in milliseconds) can also be defined.
SIP register timers: a comma separated list of sip register timers (minimum, maximum, default outgoing, timeout). Minimum and maximum timers are used to define lower and upper levels of incoming SIP registrations. The default outgoing timer is used to define the default timeout that will be requested in outgoing SIP registrations. The timeout is used to retry an outgoing register in case of failure of the previous one.
Voicemail email sender: Sender address of voicemail emails
WAV file voicemail email compression:
- Yes: attached wav file will use compression (NB: Some Windows Media Player versions do not support wav compression correctly).
- No: attached wav file won't use compression.
- Disabled: no audio file will be attached to email.
Server side DSCP marking: enter a comma separated list of 4 DSCP values for SIP, RTP audio, RTP video, IAX. Possible values are listed in the Annex. This setting should not contain any spaces and changing these require you to fully restart Communication Server.
RTP Port Ranges: A list of RTP port ranges from which to allocate ports each call. By default (when left empty), only one range is defined: 10000-20000 and its id is 0. The next range defined will have id 1 and so on. Up to 32 ranges can be defined, separated by commas. Phones and trunks will use ports from range 0 unless specified otherwise.
Number of park slots: Specify the number of park slots available. Typically this is set to 5 times the number of netConsole.
Parktime: Specifies the maximum park time in seconds. If a net.Console is used, it should be set to 3600.
In-band ringing: When set to 'Never', never generate an In-band ring tone, when set to 'yes', always generate in-band ring tone, when set to 'no' activate old behavior with out-of-band ringing followed by in-band ringing. The recommended value is 'never'. If in some case no ringing tone is heard, it can be set to 'yes'.
Group-pickup extension: set the extension to be used to pick up a ringing phone of the same pickup-group
Communication Server console timeout: set a maximum time an Communication Server console can be open. The time is to be set in seconds and its default value is one day (86400 s).
Blind transfer key: allows caller or called device to transfer ongoing call to another party (internal extension or external number). Please consider this function as very unsecure and use it at your own risks! Use option '#' for normal operation, 'custom' is special for Escaux FMU. Linked global variable: DialAsteriskOption.
Voicemail to email click-to-dial: set to enable if you want to add a link in the emails that you receive when a message is left in your voicemail. This link will transfer you to a service to hear your message.
Voicemail to email click-to-dial service (default: 8500): set the number to be called to perform the service mentionned above. Parameters available for this service are : msgcid (the caller), msgcalledid (the user called), msgid (message id in the mailbox), msgdur (message duration).
Voicemail to email click-to-dial call a phone service (default: user's extension) : indicate here the number to use to reach the voicemail owner. By default, it will call its extension. The parameters available to create the service are the same as for the previous option.
Sync Extension Policy (default: Minimum): Specify the action allowed in the context of an apply-extension-change. Set to 'Minimum' to only allow SIP reload if a phone has been added. Set to 'AllowVoicemailReload', to allow voicemail reload if an extension has been added or modified. The settings does not concern an apply-sop-change or an apply-cluster-change.
Disable IAX on external interfaces (default: no): Allows you to select if you want to bind to the IAX service to external interface. 'no' if you want to bind to localhost, 'yes' if you want to bind it to the external interface. If you do not use IAX, then you can set this option to 'yes'.
Disable AMI on external interfaces (default: yes): Allows you to select if you want to bind to the Communication Server Management Interface to external interface. 'no' if you want to bind to localhost, 'yes' if you want to bind it to the external interface. Since SOP API 4.0.0+ you can use 'yes' otherwise you must use 'no'.
Codec to use: It will create a codecs list with priority to use for the whole system.
Voicemail maximum length message (in minute): This will set the maximum length in minute of an incoming message. By default, it is set to 30 minutes.
Voicemail notification (by email) language (default : english): Allow you to configure notification language. It can be configured to send to e-mail in several languages.
Always fake user rejection: When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password. This reduces the ability of an attacker to scan for valid SIP usernames. Default is 'yes'.
Allow guest calls: Allow calls to the Default Context without any authentication. Important when listening on a public IP. Default is 'no'.
Email notification for new voicemail (default: yes): This option enable the notification sent by email when a voicemail is received. Note that a restart of Communication Server is needed to take the new value into account.
MidCallflow triggering activation key sequence: This is the key sequence used to activate the midcallflow triggering. See the action ManageMidCallflowTriggering for more information.
Use custom script for new voicemail: (default: no) Set to 'yes' if you want to use the custom script when a user receives a new voicemail, leave it to 'no' to keep the usual behavior. (This parameter is used only for Fix Mobile Unification, you should not have to change it.)
Extension assigned to the SMSService.Service callflow:
- Set the extension assigned to the SMSService.Service callflow. (This parameter is used only for Fix Mobile Unification, you should not have to change it.)
- This is only available in the version 2.x and has been deprecated in 3.x. The extension is now always vm_notify_processor in the technical context. Note that in the callflow you might want to use the GetSharedVar in order to retrieve the variable VM_USER and VM_CALLERID described below.
Phone_id to use in the externnotify script:
- (default: dp) Select the phone to notify. 'dp' stand for Desktop phone, sp for 'Soft phone'. (This parameter is used only for Fix Mobile Unification, you should not have to change it.)
- This is only available in the version 2.x and has been deprecated in 3.x. The extension vm_notify_initiator in the technical context can be used instead. In the callflow attached to this extension, it is possible to get the extension of the user in the channel variable VM_USER and the caller id of the person who let the voicemail in the variable VM_CALLERID. This callflow can ring a phone via the CallDevice action.
Voicemail extension (used for the externnotify script):
- Set the extension of the Voicemail. It will be use used as the caller_id when notifying the users. It's usually 8500 (This parameter is used only for Fix Mobile Unification, you should not have to change it)
- This is only available in the version 2.x and has been deprecated in 3.x. This can be directly handled in the callflow if needed.
Additional SIP sockets: A comma separated list of "IP Address:Port" or "IP Address:PortStart-PortEnd" on which Communication Server should listen, in addition to the main SOP IP. These can then be assigned to a SIP trunk using the field "SIP socket ID". The first one in the list will have ID 0 and so on. A full restart of Communication Server is needed when changing this.
Use external (callflow) check for Call Admission Control: (default: no) If set to 'yes', launch a special callflow which will decide to admit a call or not, instead of using an internal calculation of medialinks usage.
Debconf: See Admin Guide
IAX trunk frequency: The frequency in ms at which the IAX audio frame needs to be sent when the IAX trunk mode has been enabled. This one can be enabled in the OutgoingIAXInterface. The default value is 20 ms. The maximal value is 999 ms.
IAX register timers: A comma separated list of iax register timers in seconds in the order of minimum register time, maximum register time.. It should not contain any spaces. The default value is 60,60.
Enable enriched Remote Party Presence Information: This option enables the Remote Party Presence feature. Default is 'no'. See Remote Party Presence for more information.
Customer manager interface password: You can specify a password that you can use to connect to the manager interface. The username is always 'customer' (without the quotes).
Voicemail minimum length (in seconds): This setting can be used to eliminate messages which are shorter than a given amount of time in seconds. Default value = 2 seconds.
Voicemail maximum silence (in seconds): This setting defines how long Asterisk will wait for a contiguous period of silence before terminating an incoming call to voice mail. The default value is 10.
Max calls: Maximum number of simultaneous calls. If this value is exceeded, the new calls are rejected
Strict dialog id matching (default: yes): If set to 'yes', enable slow pedantic checking for Call-IDs, multiline SIP headers and URI-encoded headers, and enables more strict SIP RFC compliancy (available in Communication Server 2 only).
SIP Overload Control: Sip overload control allows to blacklist a known peer (ex: a sip trunk, a phone, ...) or an unknown peer (ex: a phone that hasn't been configured) if the peer sends to many invites in a certain timeframe.
- If during "polling period" seconds, there are more than "Max call per period" from a peer, the next invites will be refused by a SIP error 480 Temporarily unavailable and the "Blacklisting time after overload" (in seconds) will be initiated. If during this time, the peer sends another invite, it will be blacklisted for "IP Blacklisted time" seconds. By doing this, calls are first dropped without impact (SIP 480) and completely rejected if the peer continues sending invites. All unknown peers count as one single peer (so all invites of all unknown peers are accumulated) while it is on a per peer basis for known peers.
- The configuration should contain a comma separated list of value in following order:
  - The "Max Call per period" for the unknown peers
  - The "Polling period" for the unknown peers
  - The "Blacklisting time after overload" for the unknown peers
  - The "IP Blacklisted time" for the unknown peers
  - The default "Max Call per period" for the known peers
  - The default "Polling period" for the known peers
  - The default "Blacklisting time after overload" for the known peers
  - The default "IP Blacklisted time" for the known peers
  - where:
    - Max Call per period: The maximum number of call allowed during polling period. If the maximum is reached the SIP requests will be rejected with the a SIP error 480 Temporarily unavailable'
    - Polling period: Period during which the maximum number of calls is to be checked. At the end of each period the overload control counters are reset.
    - Blacklisting time after overload: The time after the overload during which the IP address of incoming SIP requests will be blacklisted
    - IP blacklisted time: The time during which the IP address will remain blacklisted

The "SIP Overload Control" behaviour is described in the drawing below:

Post-Install Actions

Communication Server has to be restarted after module installation. This can be done through the SOPShell :

Navigate to: Subsystems > Communication Server > Restart

Check dependencies :

In case this module is installed on a Baseline 1 High Availability SOP that is currently in standby mode, you need to use the shell plugin to deactivate the processes. This is available in the High Availability module version > 2.6.0.

Note

Be sure to use a up to date version of net.Desktop if you plan on using the connected line update feature. To find which version you need to use, refer to the release notes of net.Desktop.

Annex

Diffserv traffic class	Hex value	Dec value
CS0	0x00	0
CS1	0x08	8
CS2	0x10	16
CS3	0x18	24
CS4	0x20	32
CS5	0x28	40
CS6	0x30	48
CS7	0x38	56
AF11	0x0A	10
AF12	0x0C	12
AF13	0x0E	14
AF21	0x12	18
AF22	0x14	20
AF23	0x16	22
AF31	0x1A	26
AF32	0x1C	28
AF33	0x1E	30
AF41	0x22	34
AF42	0x24	36
AF43	0x26	38
EF	0x2E	46

Note that decimal values are accepted as well as of Communication Server 2.5.0. Note that a decimal value will span the complete Differentiated Services field. This includes the DSCP value and the ECN bits.

For more information see:

Note : This module is using a modified version of Asterisk project. Please contact D4SP Support (tac@destiny.eu) to request the source code