asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	f430d38bb0	Update CHANGES and UPGRADE.txt for 16.8-cert14	4 years ago
Joshua C. Colp	393d24db55	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. ASTERISK-29838 Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d	4 years ago
Kevin Harwell	e6ecaf292d	AST-2022-005: pjproject - undefined behavior after freeing a dialog set ASTERISK-29945 #close Change-Id: Ia8ce6d82b115c82c1138747c72a0adcaa42b718c (cherry picked from commit `7f76dce133`)	5 years ago
Kevin Harwell	aaba516fb6	AST-2022-006: pjproject - unconstrained malformed multipart SIP message ASTERISK-29945 #close Change-Id: Ic58957afc453195d53c2bd25c905df3d91d1abe6	5 years ago
Kevin Harwell	2be8154ca1	AST-2022-004: pjproject - possible integer underflow on STUN message ASTERISK-29945 #close Change-Id: I721cd254e4f8aa6d3a97a37529cca53519694c54	5 years ago
Sean Bright	7420644f6d	pbx.c: Don't remove dashes from hints on reload. When reloading dialplan, hints created dynamically would lose any dash characters. Now we ignore those dashes if we are dealing with a hint during a reload. ASTERISK-28040 #close Change-Id: I95e48f5a268efa3c6840ab69798525d3dce91636	5 years ago
George Joseph	b433fcc58c	BuildSystem: Check for alternate openssl packages OpenSSL is one of those packages that often have alternatives with later versions. For instance, CentOS/EL 7 has an openssl package at version 1.0.2 but there's an openssl11 package from the epel repository that has 1.1.1. This gets installed to /usr/include/openssl11 and /usr/lib64/openssl11. Unfortunately, the existing --with-ssl and --with-crypto ./configure options expect to point to a source tree and don't work in this situation. Also unfortunately, the checks in ./configure don't use pkg-config. In order to make this work with the existing situation, you'd have to run... ./configure --with-ssl=/usr/lib64/openssl11 \ --with-crypto=/usr/lib64/openssl11 \ CFLAGS=-I/usr/include/openssl11 BUT... those options don't get passed down to bundled pjproject so when you run make, you have to include the CFLAGS again which is a big pain. Oh... To make matters worse, although you can specify PJPROJECT_CONFIGURE_OPTS on the ./configure command line, they don't get saved so if you do a make clean, which will force a re-configure of bundled pjproject, those options don't get used. So... * In configure.ac... Since pkg-config is installed by install_prereq anyway, we now use it to check for the system openssl >= 1.1.0. If that works, great. If not, we check for the openssl11 package. If that works, great. If not, we fall back to just checking for any openssl. If pkg-config isn't installed for some reason, or --with-ssl=<dir> or --with-crypto=<dir> were specified on the ./configure command line, we fall back to the existing logic that uses AST_EXT_LIB_CHECK(). * The whole OpenSSL check process has been moved up before THIRD_PARTY_CONFIGURE(), which does the initial pjproject bundled configure, is run. This way the results of the above checks, which may result in new include or library directories, is included. * Although not strictly needed for openssl, We now save the value of PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used again if a re-configure is triggered. ASTERISK-29693 Change-Id: I341ab7603e6b156aa15a66f43675ac5029d5fbde	5 years ago
Joshua C. Colp	7b43f9f3e9	res_pjsip_nat: Don't rewrite Contact on REGISTER responses. When sending a SIP response to an incoming REGISTER request we don't want to change the Contact header as it will contain the Contacts registered to the AOR and not our own Contact URI. ASTERISK-29235 Change-Id: I35a0723545281dd01fcd5cae497baab58720478c (cherry picked from commit `d65041ebea`)	5 years ago
Asterisk Development Team	0964440c7d	Update CHANGES and UPGRADE.txt for certified/16.8-cert10	5 years ago
Kevin Harwell	7de850f476	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. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	5 years ago
Kevin Harwell	e7bf2dff2c	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. ASTERISK-29415 #close Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d	5 years ago
Asterisk Development Team	91b991f242	Update for certified/16.8-cert9	5 years ago
Joshua C. Colp	a3481f07f2	core: Don't play silence for Busy() and Congestion() applications. When using the Busy() and Congestion() applications the function ast_safe_sleep is used by wait_for_hangup to safely wait on the channel. This function may send silence if Asterisk is configured to do so using the transmit_silence option. In a scenario where an answered channel dials a Local channel either directly or through call forwarding and the Busy() or Congestion() dialplan applications were executed with the transmit_silence option enabled the busy or congestion tone would not be heard. This is because inband generation of tones (such as busy and congestion) is stopped when other audio is sent to the channel they are being played to. In the given scenario the transmit_silence option would result in silence being sent to the channel, thus stopping the inband generation. This change adds a variant of ast_safe_sleep which can be used when silence should not be played to the channel. The wait_for_hangup function has been updated to use this resulting in the tones being generated as expected. ASTERISK-29485 Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133	5 years ago
Asterisk Development Team	c3b6b056c4	Update for certified/16.8-cert8	5 years ago
Asterisk Development Team	8616ea0504	Update CHANGES and UPGRADE.txt for certified/16.8-cert6	5 years ago
George Joseph	a41edfb46f	res_pjsip_session: Make reschedule_reinvite check for NULL topologies When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	5 years ago
Joshua C. Colp	8713492758	res_rtp_asterisk: Force resync on SSRC change. When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	5 years ago
Joshua C. Colp	d1057ac427	channel: Fix crash in suppress API. There exists an inconsistency with framehook usage such that it is only on reads that the frame should be freed, not on writes as well. ASTERISK-29071 Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472	6 years ago
Joshua C. Colp	3875f7ac37	asterisk: Update copyright. ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	6 years ago
Asterisk Development Team	9e8def6e8b	Update for certified/16.8-cert7	6 years ago
Ben Ford	b9fc07ca53	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. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	6 years ago
Joshua C. Colp	54f4be115f	channel: Fix memory leak in suppress API. A frame suppression API exists as part of channels which allows audio frames to or from a channel to be dropped. The MuteAudio AMI action uses this API to perform its job. This API uses a framehook to intercept flowing audio and drop it when appropriate. It is the responsibility of the framehook to free the frame it is given if it changes the frame. The suppression API failed to do this resulting in a leak of audio frames. This change adds the freeing of these frames. ASTERISK-29071 Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b	6 years ago
Joshua C. Colp	df34b007d9	bridge: Don't try to match audio formats. When bridging channels we were trying to match the audio formats of both sides in combination with the configured formats. While this is allowed in SDP in practice this causes extra reinvites and problems. This change ensures that audio streams use the formats of the first existing active audio stream. It is only when other stream types (like video) exist that this will result in re-negotiation occurring for those streams only. ASTERISK-28871 Change-Id: I22f5a3e7db29e00c165e74d05d10856f6086fe47	6 years ago
Ben Ford	b12fcaaaf0	res_pjsip_session.c: Check topology on re-invite. Removes an unnecessary check for the conditional that compares the stream topologies to see if they are equal to suppress re-invites. This was a problem when a Digium phone received an INVITE that offered codecs different than what it supported, causing Asterisk to send the re-invite. ASTERISK-29303 Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63	6 years ago
Asterisk Development Team	5d0a019e13	Update for certified/16.8-cert6	6 years ago
Kevin Harwell	bb5b336f78	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. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	6 years ago
Alexander Traud	93468c531a	rtp: Enable srtp replay protection Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	6 years ago
Sean Bright	07eddbd56a	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	6 years ago
Joshua C. Colp	23e96e6e17	pjsip: Make modify_local_offer2 tolerate previous failed SDP. If a remote side is broken and sends an SDP that can not be negotiated the call will be torn down but there is a window where a second 183 Session Progress or 200 OK that is forked can be received that also attempts to negotiate SDP. Since the code marked the SDP negotiation as being done and complete prior to this it assumes that there is an active local and remote SDP which it can modify, while in fact there is not as the SDP did not successfully negotiate. Since there is no local or remote SDP a crash occurs. This patch changes the pjmedia_sdp_neg_modify_local_offer2 function to no longer assume that a previous SDP negotiation was successful. ASTERISK-29196 Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e	6 years ago
Joshua C. Colp	6b5306ecc5	res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent. This adds support for both Digium and Sangoma user agent strings for the Sangoma specific body supplement. Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482 (cherry picked from commit `7e4bb4ed11`)	6 years ago
Asterisk Development Team	10b274d1ab	Update for certified/16.8-cert5	6 years ago
Kevin Harwell	81b48f2d10	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. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	6 years ago
Ben Ford	6057050899	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. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	6 years ago
Asterisk Development Team	21635f1e40	Update for certified/16.8-cert4	6 years ago
Asterisk Development Team	5e146dde81	Update CHANGES and UPGRADE.txt for certified/16.8-cert4	6 years ago
Asterisk Development Team	f777a0d879	Update for certified/16.8-cert4-rc4	6 years ago
George Joseph	0757a5113d	Revert "Revert "Revert "core_unreal / core_local: Add multistream and re-negotiation.""" First it was in. Then it was out. Then it was in again. Now it's out again because of a faxing issue discovered by the swvx team. Reason for revert: Faxing issue Change-Id: Ia417853130361b69f83f70bbc799493bb9f2fe27	6 years ago
Asterisk Development Team	cb9f1759fb	Update for certified/16.8-cert4-rc3	6 years ago
Joshua C. Colp	d5df3b95c8	res_pjsip_session: Fix stream name memory leak. When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	6 years ago
Joshua C. Colp	3e54af3afb	res_pjsip_session: Fix session reference leak. The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	6 years ago
George Joseph	f74ecc4498	logger.h: Fix ast_trace to respect scope_level ast_trace() was always emitting messages when it's level was set to -1 because it was ignoring scope_level. Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506	6 years ago
George Joseph	251c9a1891	bridge_softmix/sfu_topologies_on_join: Ignore topology change failures When a channel joins a bridge, we do topology change requests on all existing channels to add the new participant to them. However the announcer channel will return an error because it doesn't support topology in the first place. Unfortunately, there doesn't seem to be a reliable way to tell if the error is expected or not so the error is ignored for all channels. If the request fails on a "real" channel, that channel just won't get the new participant's video. Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0	6 years ago
George Joseph	1620e94af5	res_pjsip_session: Fix issue with COLP and 491 The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	6 years ago
George Joseph	f67c58aa13	debugging: Add enough to choke a mule Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	6 years ago
George Joseph	9e54db24b6	res_pjsip_session: Handle multi-stream re-invites better When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	6 years ago
Ben Ford	c2701d8069	Bridging: Use a ref to bridge_channel's channel to prevent crash. There's a race condition with bridging where a bridge can be torn down causing the bridge_channel's ast_channel to become NULL when it's still needed. This particular case happened with attended transfers, but the crash occurred when trying to publish a stasis message. Now, the bridge_channel is locked, a ref to the ast_channel is obtained, and that ref is passed down the chain. Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814	6 years ago
George Joseph	fb78adce21	stream.c: Added 2 more debugging utils and added pos to stream string * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros which are shortcuts for ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP)) * Added the stream position to the string representation of the stream. * Fixed some formatting in ast_stream_to_str(). Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b	6 years ago
George Joseph	d942fe4837	scope_trace: Added debug messages and added additional macros The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages at the same level as the scope level. This allows the same messages to be printed to the debug log when AST_DEVMODE isn't enabled. Also added a few variants of the SCOPE_EXIT macros that will also call ast_log instead of ast_debug to make it easier to use scope tracing and still print error messages. Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21	6 years ago
George Joseph	0430e1b3c3	logger.c: Added a new log formatter called "plain" Added a new log formatter called "plain" that always prints file, function and line number if available (even for verbose messages) and never prints color control characters. It also doesn't apply any special formatting for verbose messages. Most suitable for file output but can be used for other channels as well. You use it in logger.conf like so: debug => [plain]debug console => [plain]error,warning,debug,notice,pjsip_history messages => [plain]warning,error,verbose Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d	6 years ago
George Joseph	950784a446	ast_coredumper: Fix issues with naming If you run ast_coredumper --tarball-coredumps in the same directory as the actual coredump, tar can fail because the link to the actual coredump becomes recursive. The resulting tarball will have everything _except_ the coredump (which is usually what you need) There's also an issue that the directory name in the tarball is the same as the coredump so if you extract the tarball the directory it creates will overwrite the coredump. So: * Made the link to the coredump use the absolute path to the file instead of a relative one. This prevents the recursive link and allows tar to add the coredump. * The tarballed directory is now named <coredump>.output instead of just <coredump> so if you expand the tarball it won't overwrite the coredump. Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea	6 years ago

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