asterisk 16.16.2

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

16.16.206 Mar 2021 01:45 minor feature: AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. When Asterisk sends a reinvite negotiating T38 faxing, it's possible a. Crash can occur if the response contains a m=image and zero port. The Reinvite callback code now checks session_media to see if it is null or Not before trying to access the udptl variable on it.
16.16.119 Feb 2021 20:25 minor feature: AST-2021-002: Remote crash possible when negotiating T.38 When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for. The initial invite the re-invite gets delayed. When Asterisk does Finally send the re-inivite the SDP includes streams for both audio And T.38. This happens because when the pending topology and active topologies. Differ (pending stream is not in the active) in the delayed scenario The pending stream is appended to the active topology. However, in The fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice. Versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38. And the endpoint responded with a declined T.38 stream then Asterisk Would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a. Valid fax session (associated udptl object) before changing the T.38 state internally. rtp: Enable srtp replay protection. Add option "srtpreplayprotection" rtp.conf to enable srtp. Replay protection. res_pjsip_diversion: adding more than one histinfo to Supported New responses sent within a PJSIP sessions are based on those that were. Sent before. Therefore, adding/modifying a header once causes it to be Sent on all responses that follow. Sending 181 Call Is Being Forwarded many times first adds "histinfo". Duplicated more and more, and eventually overflows past the array Boundary. This commit adds a check preventing adding "histinfo" more than once. And skipping it if there is no more space in the header. Similar overflow situations can also occur in res_pjsip_path and. Res_pjsip_outbound_registration so those were also modified to Check the bounds and suppress duplicate Supported values. res_rtp_asterisk.c: signed mismatch that leads to overflow pjsip:
16.15.124 Dec 2020 11:25 minor feature: Update for 16.15.1 res/res_pjsip_diversion: prevent crash on tel: uri in History-Info. Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter.
16.14.106 Nov 2020 11:45 minor feature: AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. If Asterisk sends out an INVITE and receives a challenge with a. Different nonce value each time, it will continuously send out INVITEs, Even if the call is hung up. The endpoint must be configured for Outbound authentication for this to occur. A limit has been set on Outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. AST-2020-001 - res_pjsip: Return dialog locked and referenced. Pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference. And removes the lock prior to returning. This makes it possible, Under some circumstances, for another thread to free said dialog Before the thread that created it attempts to use it again. Of Course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created. Dialog both locked, and with an added reference. This allows the Caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now. Held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects. Like the transaction, to disappear. For example if there is a TCP Transport error.
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.
16.6.222 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.
16.5.106 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.
16.4.112 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.
16.2.102 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.
16.1.127 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.
16.0.116 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.