32516 Commits (certified/16.8-cert9)

 

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Author	SHA1	Message	Date
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	6 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
Joshua C. Colp	7b1ace9040	res_pjsip_session: Don't aggressively terminate on failed re-INVITE. ... Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503	6 years ago
George Joseph	d189bf9ae5	res_pjsip_session: Ensure reused streams have correct bundle group ... When a bundled stream is removed, its bundle_group is reset to -1. If that stream is later reused, the bundle parameters on session media need to be reset correctly it could mistakenly be rebundled with a stream that was removed and never reused. Since the removed stream has no rtp instance, a crash will result. Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7	6 years ago
Joshua C. Colp	b8bb63c82f	pjsip: Add timer refactor patch and timer 0 cancellation. ... There have been numerous timer issues over the years which resulted in Teluu doing a major refactor of the timer implementation whereby the timer entries themselves are not trusted. This change backports this refactor which has been shown to resolve timer crashes. Additionally another patch has been backported to prevent timer entries with an internal id of 0 from being canceled. This would result in the invalid timer id of 0 being placed into the timer heap causing issues. This is also a backport. Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7 (cherry picked from commit 9ce6d46aea)	6 years ago
Kevin Harwell	5e58fc3939	manager - Add Content-Type parameter to the SendText action ... This patch allows a user of AMI to now specify the type of message content contained within by setting the 'Content-Type' parameter. Note, the AMI version has been bumped for this change. ASTERISK-28945 #close Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb	6 years ago
cmaj	4e9e503132	Makefile: Fix certified version numbers ... Adds sed before awk to produce reasonable ASTERISKVERSIONNUM on certified versions of Asterisk eg. 16.8-cert3 is 160803 instead of the previous 00800. ASTERISK-29021 #close Change-Id: Icf241df0ff6db09011b8c936a317a84b0b634e16	6 years ago
George Joseph	8ebe4c772e	Scope Trace: Make it easier to trace through synchronous tasks ... Tracing through synchronous tasks was a little troublesome because the new thread's stack counter reset to 0. This change allows a synchronous task to set its trace level to be the same as the thread that pushed the task. For now, the task's level has to be passed in the task's data structure but a future enhancement to the taskprocessor subsystem could automatically set the trace level of the servant to be that of the caller. This doesn't really make sense for async tasks because you never know when they're going to run anyway. Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825	6 years ago
George Joseph	8076ed59f0	scope_trace: Add/update utilities ... * Added a AST_STREAM_STATE_END sentinel * Add ast_stream_to_str() * Add ast_stream_state_to_str() * Add ast_stream_get_format_count() * Add ast_stream_topology_to_str() * Add ast_stream_topology_get_active_count() * Add ast_format_cap_append_names() * Add ast_sip_session_get_name() Change-Id: I132eb5971ea41509c660f64e9113cda8c9013b0b	6 years ago
George Joseph	9d3b01e874	Scope Trace: Add some new tracing macros and an ast_str helper ... Created new SCOPE_ functions that don't depend on RAII_VAR. Besides generating less code, the use of the explicit SCOPE_EXIT macros capture the line number where the scope exited. The RAII_VAR versions can't do that. * SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use RAII_VAR and therefore needs needs one of... * SCOPE_EXIT(...): Decrements the trace stack counter and optionally prints a message. * SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter, optionally prints a message, then executes the expression. SCOPE_EXIT_EXPR(break, "My while got broken\n"); * SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter, optionally prints a message, then returns without a value. SCOPE_EXIT_RTN("Bye\n"); * SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace stack counter, optionally prints a message, then returns the value specified. SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc); Create an ast_str helper ast_str_tmp() that allocates a temporary ast_str that can be passed to a function that needs it, then frees it. This makes using the above macros easier. Example: SCOPE_ENTER(1, Format Caps 1: %s Format Caps 2: %s\n", ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP), ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP)); The calls to ast_str_tmp create an ast_str of the specified initial length which can be referenced as STR_TMP. It then calls the expression, which must return a char *, ast_strdupa's it, frees STR_TMP, then returns the ast_strdupa'd string. That string is freed when the function returns. Change-Id: I44059b20d55a889aa91440d2f8a590865998be51 (cherry picked from commit 43ba72dea0)	6 years ago
George Joseph	360efdd94f	Scope Tracing: A new facility for tracing scope enter/exit ... What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027 (cherry picked from commit 6a0c472374)	6 years ago
George Joseph	944d691319	frame.c: Make debugging easier ... * ast_frame_subclass2str() and ast_frame_type2str() now return a pointer to the buffer that was passed in instead of void. This makes it easier to use these functions inline in printf-style debugging statements. * Added many missing control frame entries in ast_frame_subclass2str. Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc (cherry picked from commit fdcb3e2ead)	6 years ago
Asterisk Development Team	017416381f	Update for certified/16.8-cert4-rc2	6 years ago