|Tags||communication conferencing telephony sip|
16.11.120 Jun 2020 22:25 minor feature: Res_ari: create channel request channelId parameter parsing If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. it to parse the channelId, other_channel_id parameter correclty.
16.11.012 Jun 2020 21:25 minor feature:
16.10.001 May 2020 22:05 minor feature:
16.9.013 Mar 2020 16:25 minor feature:
16.8.005 Feb 2020 20:25 minor feature:
16.7.024 Dec 2019 10:05 minor feature:
16.6.222 Nov 2019 19:25 minor feature: Update CHANGES and UPGRADE.txt for 16.6.2 manager.c: Prevent the Originate action from running the Originate app. If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned. If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option.
16.6.117 Oct 2019 16:45 minor feature: Pjproject_bundled: Replace earlier reverts with official. in pjproject 2.9 caused us to revert some of their changes as a work around. This introduced another where pjproject. Wouldn't build with older gcc versions such as that found on CentOS 6. This commit replaces the reverts with the official. For the original and allows pjproject to be built on CentOS 6 again. res_pjsip_mwi: potential double unref, and potential unwanted double link. When creating an unsolicited MWI aggregate subscription it was possible for. The subscription object to be double unref'ed. This patch removes the explicit Unref as it is not needed since the RAII_VAR will handle it at function end. Less concerning there was also a that could potentially allow the aggregate. Subscription object to be added to the unsolicited container twice. This patch Ensures it is added only once.
16.6.010 Oct 2019 02:25 minor feature:
16.5.106 Sep 2019 22:25 minor feature: AST-2019-005 - translate: Don't assume all frames will have a src. This change removes the assumption that a frame will always have a src set on it. This assumption is incorrect. Given a scenario where an RTP packet is received with no payload. The resulting audio frame will have no samples. If this frame goes Through a signed linear translation path an interpolated frame can be created (if generic packet loss concealment is enabled) that has. Minimal data on it, including no src. If this frame is given to a Translation path a crash will occur due to the lack of src. AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media After receiving a 200 OK with a declined stream in response to a T.38. Initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before. Attempting to use it.
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.
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