asterisk 16.5.0

Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source.

Tags communication conferencing telephony sip
License GNU GPL
State stable

Recent Releases

16.5.026 Jul 2019 10:45 minor feature:
16.4.112 Jul 2019 17:45 minor feature: Res_pjsip_messaging: Check for body in in-dialog message We now check that a body exists and it has a length 0 before. Attempting to process it. chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media. Stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional. Media stream with no joint codec a crash will occur as the code Assumes that at least one joint codec will exist in this Scenario. This change removes this assumption.
16.4.031 May 2019 13:45 minor feature:
16.3.006 Apr 2019 08:25 minor feature:
16.2.102 Mar 2019 03:25 minor feature: Res_pjsip_sdp_rtp: return code from apply_negotiated_sdp_stream Apply_negotiated_sdp_stream was returning a "1" when no joint. Capabilities were found on an outgoing call instead of a "-1". This indicated to res_pjsip_session that the handler DID handle. The sdp when in fact it didn't. Without the appropriate setup, a subsequent media frame coming in would have an invalid stream_num. And cause a seg fault when the stream was attempted to be retrieved. Apply_negotiated_sdp_stream now returns the correct "-1" and any. Media is now discarded before it reaches the core stream processing. CI: Update jenkinsfiles with new Gerrit URLs The recent upgrade of Gerrit to 2.16 elimiated referencing a. Repository in a way the jenkinsfiles were relying on so The URL references were changed to a more consistent and supported Format.
16.2.016 Feb 2019 09:05 minor feature:
16.1.127 Dec 2018 23:05 minor feature: Revert "stasis_cache: Stop caching stasis subscription change messages" This commit caused with polling when combined with the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis". This reverts commit 17d6d9e1e7d0db04ebd8d2e0cd9e087ec5462e2f.
16.1.013 Dec 2018 06:45 minor feature:
16.0.116 Nov 2018 21:45 minor feature: AST-2018-010: length of buffer needed for SRV and NAPTR results When dn_expand was being called on SRV and NAPTR results, the. Return value was being used to calculate the size of the buffer Needed to store the host names. Since dn_expand returns the Length of the COMPRESSED name the buffer could be too short to hold the EXPANDED name. The expanded name is NULL terminated so using strlen() is the correct way to determine the length. Actually needed for the buffer.
16.0.010 Oct 2018 09:32 major feature: Improved Video Conferencing Performance Asterisk 16 builds upon the extensive video conferencing capabilities introduced in Asterisk 15 to provide a dramatically improved video experience for users. Asterisk now delivers superior video performance for all network conditions, which reduces the risk of frozen video frames and provides a world-class framework for creating cutting-edge video applications. New Text-Based Data Capabilities Support for Enhanced Messaging has been added to give developers the ability to build rich client applications with text-based data exchanges. Now, multi-party video conferencing client applications can share URLs, list conference participants, highlight talkers, and enable multi-party chat. Improved Call Handling Asterisk 16 has also undergone significant performance enhancements to better handle SIP calling by decreasing the system memory and CPU consumption required during high volume situations, most notably when utilizing the PJSIP channel driver.