asterisk 16.30.0

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

16.30.013 Jan 2023 10:45 minor feature:
16.29.102 Dec 2022 16:25 minor feature: Update for 16.29.1 pjproject: 2.13 security Backports two security (c4d3498 and 450baca) from pjproject 2.13. pjsip_transport_events: possible use after free on transport. It was possible for a module that registered for transport monitor. Events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the. Monitor for the transport. The is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we. Create monitors for them, then decrement the count when the Transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip. Address and port. This way the module needing to monitor the Transport doesn't have to hold on to the transport object itself to Unregister. It just has to save the transport_key. Added the pjsip_transport reference increment and decrement. Changed the internal transport monitor container key from the. Transport- obj_name (which may not be unique anyway) to the Transport_key. Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that. Fills a buffer with the transport_key using a passed-in Pjsip_transport. Added the following functions: Ast_sip_transport_monitor_register_key Ast_sip_transport_monitor_register_replace_key Ast_sip_transport_monitor_unregister_key And marked their non-key counterparts as deprecated. Updated res_pjsip_pubsub and res_pjsip_outbound_register to use. The new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor. Functionality but doesn't have a persistent object other than Contact to store a transport key. At this time, it continues to Use the non-key monitor functions. manager: prevent file access outside of config dir Add live_dangerously flag to manager and use this flag to. Determine if a configuation file outside of AST_CONFIG_DIR Should be read. ooh323c: not checking for IE minimum le
16.29.020 Oct 2022 13:45 minor feature:
16.28.019 Aug 2022 11:05 minor feature:
16.27.024 Jun 2022 07:45 minor feature:
16.26.122 May 2022 04:45 minor feature: Res_pjsip_transport_websocket: Also set the remote name. As part of PJSIP 2.11 a behavior change was done to require a matching remote hostname on an established transport for. Secure transports. Since the Websocket transport is considered a secure transport this caused the existing connection to not be found and used. We now set the remote hostname and the transport can be found.
16.26.013 May 2022 04:05 minor feature:
16.25.327 Apr 2022 14:45 minor feature: Res_pjsip_stir_shaken.c: enabled when not configured. There was an with the conditional where STIR/SHAKEN would be. Enabled even when not configured. It has been changed to ensure that if a profile does not exist and stir_shaken is not set in pjsip.conf, then. The conditional will return from the function without performing STIR/SHAKEN operations.
16.25.215 Apr 2022 14:45 minor feature: Doing a fresh summary Update for 16.25.2. Update CHANGES and UPGRADE.txt for 16.25.2. Update CHANGES and UPGRADE.txt for 16.25.2. func_odbc: Add SQL_ESC_BACKSLASHES dialplan function. Some databases depending on their configuration using backslashes. For escaping. When combined with the use of ' this can result in a broken func_odbc query. This change adds a SQL_ESC_BACKSLASHES dialplan function which can be used to escape the backslashes. This is done as a dialplan function instead of being always done as some databases do not require this, and always doing it would. Result in incorrect data being put into the database. AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. Adds a new configuration option, stir_shaken_profile, in pjsip.conf that. Can be specified on a per endpoint basis. This option will reference a Stir_shaken_profile that can be configured in stir_shaken.conf. The type of this option must be 'profile'. The stir_shaken option can be. Specified on this object with the same values as before (attest, verify, on), but it cannot be off since having the profile itself implies wanting STIR/SHAKEN support. You can also specify an ACL from acl.conf (along. With permit and deny lines in the object itself) that will be used to Limit what interfaces Asterisk will attempt to retrieve information from When reading the Identity header. AST-2022-001 - res_stir_shaken/curl: Limit file size and check start. Put checks in place to limit how much we will actually download, as well as a check for the data we receive at the start to ensure it begins with. What we would expect a certificate to begin with.
16.25.103 Apr 2022 01:05 minor feature: Makefile: Disable XML doc validation Make_xml_documentation was being called with the --validate. Flag set when it shouldn't have been. This was causing Build failures if neither xmllint nor xmlstarlet were installed. The correct behavior is to simply print a message that either. One of those tools should be installed for validation and Continue with the build. make_xml_documentation: Remove usage of get_sourceable_makeopts Get_sourceable_makeopts wasn't handling variables with embedded. Double quotes in them very well. One example was the DOWNLOAD Variable when curl was being used instead of wget. Rather than Trying to get_sourceable_makeopts, it's just been removed.
16.25.025 Mar 2022 07:25 minor feature:
16.24.105 Mar 2022 18:25 minor feature: AST-2022-006: pjproject - unconstrained malformed multipart SIP message AST-2022-005: pjproject - undefined behavior after freeing a dialog set. AST-2022-004: pjproject - possible integer underflow on STUN message.
16.24.011 Feb 2022 03:45 minor feature:
16.23.010 Dec 2021 16:45 minor feature:
16.22.003 Nov 2021 07:25 minor feature:
16.21.116 Oct 2021 16:05 minor feature: Pbx_ael: crash and lockup regarding 'ael reload' Avoid infinite recursion and crash.
16.21.008 Oct 2021 11:25 minor feature:
16.20.013 Aug 2021 16:05 minor feature:
16.19.124 Jul 2021 19:05 minor feature: AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS If an SSL socket parent/listener was destroyed during the handshake. Depending on timing, it was possible for the handling callback to Attempt access of it after the fact thus causing a crash. AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. If a re-INVITE is received after we have sent a BYE request then it is possible for no channel to be present on the session. If this. Occurs we allow PJSIP to produce the offer instead. Since the call is being hung up if it produces an incorrect offer it doesn't. Actually matter. This also ensures that code which produces SDP Does not need to handle if a channel is not present. AST-2021-008 - chan_iax2: remote crash on unsupported media format If chan_iax2 received a packet with an unsupported media format, for. Example vp9, then it would set the frame's format to NULL. This could Then result in a crash later when an attempt was made to access the Format. This patch makes it so chan_iax2 now ignores/drops frames received. With unsupported media format types.
16.19.025 Jun 2021 23:45 minor feature:
16.18.007 May 2021 19:45 minor feature:
16.17.026 Mar 2021 11:25 minor feature:
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:
16.16.023 Jan 2021 05:25 minor feature:
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.
16.15.020 Nov 2020 11:25 minor feature:
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.
16.14.021 Oct 2020 22:45 minor feature:
16.13.005 Sep 2020 08:25 minor feature:
16.12.018 Jul 2020 06:45 minor feature:
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.