asterisk

Commit Graph

Author	SHA1	Message	Date
Ben Ford	5fbed5af24	res_stir_shaken: Add stir_shaken option and general improvements. Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	5 years ago
George Joseph	e88beedd08	res_pjsip_session: Fix segv in session_on_rx_response session_on_rx_response wasn't checking for a NULL dialog before attempting to get the invite session from it. Change-Id: Id13534375966cc2eb7f2b55717c9813c63c10065	5 years ago
Walter Doekes	312c23b0e1	app_queue: (Breaking change) shared_lastcall and autofill default to no If your queues.conf had _no_ [general] section, they would default to 'yes'. Now, they always default to 'no'. (Actually, commit `ed615afb7e` already partially fixed it for shared_lastcall.) ASTERISK-28951 Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6	5 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
George Joseph	2d22e34206	ACN: res_pjsip endpoint options This commit adds the endpoint options required to control Advanced Codec Negotiation. incoming_offer_codec_prefs outgoing_offer_codec_prefs incoming_answer_codec_prefs outgoing_answer_codec_prefs The documentation may need tweaking and some additional edits added, especially for the "answer" prefs. That'll be handled when things finalize. This commit is safe to merge as it doens't alter any existing functionality nor does it alter the previous codec negotiation work which may now be obsolete. Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11	5 years ago
sungtae kim	81b5e4a73f	res_pjsip.c: Added disable_rport option for pjsip.conf Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	5 years ago
George Joseph	d093e44b1e	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	5 years ago
George Joseph	955b7b4fdb	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	5 years ago
Nickolay Shmyrev	7163efd934	res_http_websocket.c: Continue reading after ping/pong Do not return error if the client received ping frame while looking for a string and just wait for another frame. ASTERISK-28958 #close Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922	5 years ago
Kevin Harwell	4eba6b9eb2	PJSIP_MEDIA_OFFER: override configuration on refresh When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	5 years ago
Kevin Harwell	cfed0ea033	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	5 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	5 years ago
George Joseph	7440fd0397	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	5 years ago
Joshua C. Colp	4f86118bd8	res_pjsip: Apply AOR outbound proxy to static contacts. The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	5 years ago
Joshua C. Colp	9b5042433b	menuselect: Resolve infinite loop in dependency scenario. Given a scenario where a module has a dependency on both an external library and a module if the external library was available and the module was not an infinite loop would occur. This happened due to the code changing the dependecy status to no failure on each dependency checking loop iteration, resulting in the code thinking that it had gone from no failure to failure each time triggering another dependency check. This change makes it so that the old dependency status is preserved throughout the dependency checking allowing it to determine that after the first iteration the dependency status does not transition from no failure to failure. ASTERISK-28930 Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84	5 years ago
Frederic LE FOLL	a423f935c9	chan_sip: chan_sip does not process 400 response to an INVITE. chan_sip handle_response() function, for a 400 response to an INVITE, calls handle_response_invite() and does not generate ACK. handle_response_invite() does not recognize 400 response and has no default response processing for unexpected responses, thus it does not generate ACK either. The ACK on response repetition comes from handle_response() mechanism "We must re-send ACKs to re-transmitted final responses". According to code history, 400 response specific processing was introduced with commit "channels/chan_sip: Add improved support for 4xx error codes" This commit added support for : - 400/414/493 in handle_response_subscribe() handle_response_register() and handle_response(). - 414/493 only in handle_response_invite(). This fix adds 400 response support in handle_response_invite(). ASTERISK-28957 Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad	5 years ago
Kevin Harwell	8b925fbda3	chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ A patch made a reference to the PJSIP_SC_NULL enumeration value, which was added to pjproject 2.8 and above thus making it so Asterisk would fail to compile with prior versions of pjproject. This patch removes the reference, and instead initializes the value to '0'. ASTERISK-28886 #close Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7	5 years ago
Università di Bologna - CESIA VoIP	0c1c386634	res_corosync: Fix crash in huge distributed environment. 1) Fix memory-leaks Added code to release ast_events extracted from corosync and stasis messages 2) Clean stasis cache when a member of the corosync cluster leaves the group Added code to remove from the stasis cache of the members remained on the group all the messages with the EID of the left member. If the device states of the left member remain in the stasis cache of other members, they will not be updated anymore and high priority cached values, like BUSY, will take precedence over current device states. 3) Stop corosync event propagation when node is not joined to the group Updated dispatch_thread_handler code to detect when asterisk is not joined to the corosync group and added some condition in publish_event_to_corosync code to send corosync messages only when joined. When a node is not joined its corosync daemon can't send messages: the cpg_mcast_joined function append new messages to the FIFO buffer until it's full and then it blocks indefinitely. In this scenario if the stasis_message_cb callback, registered by res_corosync to handle stasis messages, try to send a corosync messages, the thread of the stasis thread-pool will be blocked until the node join the corosync cluster. ASTERISK-28888 Reported by: Università di Bologna - CESIA VoIP Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2	5 years ago
Moises Silva	9445dac43b	res_http_websocket: Add payload masking to the websocket client ASTERISK-28949 Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca	5 years ago
Joshua C. Colp	00a52b4752	app_stream_echo: Fix state of added streams. When stream support was added to Asterisk the stream state was used inconsistently, resulting in odd behavior. This was then standardized to be the state of a stream from the perspective of Asterisk. This change updates the StreamEcho dialplan application to use the correct state, send only, since we are only sending to the endpoint and not expecting them to send us multiple video streams. ASTERISK-28954 Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56	5 years ago
Guido Falsi	d88e230037	chan_dadhi: Fix setvar in dahdi channels The change to how setvar works for various channels performed in ASTERISK~23756 missed some required change in the dahdi channel, where the variables are actually set while reading configuration. This change should fix the issue. ASTERISK-28955 Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274	5 years ago
Joshua C. Colp	ee8ea9275f	res_pjsip_session: Preserve label on incoming re-INVITE. When a re-INVITE is received we create a new set of streams that are then swapped in as the active streams. We did not preserve the SDP label from the previous streams, resulting in the label getting lost. This change ensures that if an SDP label is present on the previous stream then it is set on the new stream. ASTERISK-28953 Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445	5 years ago
Joshua C. Colp	a143c3a7b7	res_sorcery_memory_cache: Disallow per-object expire with full backend. The AMI action and CLI command did not take into account the properties of full backend caching. This resulted in an expired object remaining removed until a full backend update occurred, instead of having the object updated when needed. This change makes it so that the AMI action and CLI command for object expire will now fail instead of putting the cache into an undesired state. If full backend caching is enabled then only operations which act on the entire cache are available. ASTERISK-28942 Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4	5 years ago
Ben Ford	1274117102	res_stir_shaken: Add outbound INVITE support. Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is sent, the caller ID will be checked to see if there is a certificate that corresponds to it. If so, that information will be retrieved and an Identity header will be added to the SIP message. The format is: header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken Header, payload, and signature are all BASE64 encoded. The public key URL is retrieved from the certificate. Currently the algorithm and ppt are ES256 and shaken, respectively. This message is signed and can be used for verification on the receiving end. Two new configuration options have been added to the certificate object: attestation and origid. The attestation is required and must be A, B, or C. origid is the origination identifier. A new utility function has been added as well that takes a string, allocates space, BASE64 encodes it, then returns it, eliminating the need to calculate the size yourself. Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4	5 years ago
Walter Doekes	db012e8cc6	app_queue: Remove stale code in try_calling Because ring_entry() is not called, outgoing->chan is not touched here either. ASTERISK-28950 ASTERISK-28644 Change-Id: I564613715dfaf45af868251eb75a451f512af90f	5 years ago
Walter Doekes	f1cfd54976	res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h" Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78	5 years ago
Walter Doekes	0fb6738314	app_queue: Read latest wrapuptime instead of (possibly stale) copy Before this changeset, it was possible that a queue member (agent) was called even though they just got out of a call, and wrapuptime seconds hadn't passed yet. This could happen if a member ended a call _between_ a new call attempt and asterisk trying that particular member for a new call. In that case, Asterisk would check the hangup time of the call-before-the-last-call instead of the hangup time of the-last-call. ASTERISK-28952 Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3	5 years ago
Kevin Harwell	415b55af5a	pjproject: Upgrade bundled version to pjproject 2.10 This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81	5 years ago
Joshua C. Colp	de2813cf23	core_unreal / core_local: Add multistream and re-negotiation. When requesting a Local channel the requested stream topology or a converted stream topology will now be placed onto the resulting channels. Frames written in on streams will now also preserve the stream identifier as they are queued on the opposite channel. Finally when a stream topology change is requested it is immediately accepted and reflected on both channels. Each channel also receives a queued frame to indicate that the topology has changed. ASTERISK-28938 Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186	5 years ago
sungtae kim	bbe0f2230d	res_ari: Fix create channel request channelId parameter parsing If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. Fixed it to parse the channelId, other_channel_id parameter correclty. ASTERISK-28948 Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157	5 years ago
Joshua C. Colp	c84d962eae	res_rtp_asterisk: Don't assume setting retrans props means to enable. The "value" passed in when setting an RTP property determines whether it should be enabled or disabled. The RTP send and receive retrans props did not examine this to know if the buffers should be enabled. They assumed they always should be. This change makes it so that the "value" passed in is respected. ASTERISK-28939 Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc	5 years ago
Joshua C. Colp	8ad06394c4	bridge_softmix: Add additional old states for adding new source. There are three states that an old stream can be in to allow becoming a source stream in a new stream: 1. Removed 2. Inactive 3. Sendonly This change adds the two missing ones, inactive and sendonly, so if a stream transitions from those to a state where they are providing video to Asterisk we properly re-negotiate the other participants. ASTERISK-28944 Change-Id: Id8256b9b254b403411586284bbaedbf50452de01	5 years ago
George Joseph	41f3a7da4d	res_fax: Don't start a gateway if either channel is hung up When fax_gateway_framehook is called and a gateway hasn't already been started, the framehook gets the t38 state for both the current channel and the peer. That call trickles down to the channel driver which determines the state. If either channel is hung up (or in the process of being hung up), the channel driver's tech_pvt is going to be NULL which, in the case of chan_pjsip, will cause a segfault. * Added a hangup check for both the channel and peer channel before starting a fax gateway. * Added a check for NULL tech_pvt to chan_pjsip_queryoption so we don't attempt to reference a tech_pvt that's already gone. ASTERISK-28923 Reported by: Yury Kirsanov Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c	5 years ago
George Joseph	b9f42a717e	app_confbridge: Plug ref leak of bridge channel with send_events When send_events is enabled for a user, we were leaking a reference to the bridge channel in confbridge_manager.c:send_message(). This also caused the bridge snapshot to not be destroyed. Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97	5 years ago
Kevin Harwell	3d1bf3c537	Compiler fixes for gcc 10 This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	5 years ago
Ben Ford	559fa0e89c	cli.c: Fix compiler error. Added default variable value to fix a compiler error. Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553	5 years ago
sungtae kim	fa7c69f40f	res_ari: Fix create request body parameter parsing. If parameters were passed in the body as JSON to the create route they were not being parsed before checking to ensure that required fields were set. This change moves the parsing so it occurs before checking. ASTERISK-28940 Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf	5 years ago
Walter Doekes	e74dde5100	pjsip: Prevent invalid memory access when attempting to contact a non-sip URI You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri, without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S). ASTERISK-28936 Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a	5 years ago
Ben Ford	3927f79cb5	res_stir_shaken: Add inbound INVITE support. Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an INVITE, the Identity header is retrieved, parsing the message to verify the signature. If any of the parsing fails, AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this caller ID. If verification itself fails, AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in the payload does not line up with the SIP signaling, AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the verification process. A new config option has been added to the general section for stir_shaken.conf. "signature_timeout" is the amount of time a signature will be considered valid. If an INVITE is received and the amount of time between when it was received and when it was signed is greater than signature_timeout, verification will fail. Some changes were also made to signing and verification. There was an error where the whole JSON string was being signed rather than the header combined with the payload. This has been changed to sign the correct thing. Verification has been changed to do this as well, and the unit tests have been updated to reflect these changes. A couple of utility functions have also been added. One decodes a BASE64 string and returns the decoded string, doing all the length calculations for you. The other retrieves a string value from a header in a rdata object. Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913	5 years ago
Joshua C. Colp	1fcb6b1b21	bridge_channel: Don't queue unmapped frames. If a frame is written to a channel in a bridge we would normally queue this frame up and the channel thread would then act upon it. If this frame had no stream mapping on the channel it would then be discarded. This change adds a check before the queueing occurs to determine if a mapping exists. If it does not exist then the frame is not even queued at all. This stops a frame duplication from happening and from the channel thread having to wake up and deal with it. Change-Id: I17189b9b1dec45fc7e4490e8081d444a25a00bda	5 years ago
Joshua C. Colp	d2500c6273	res_fax: Don't consume frames given to fax gateway on write. In a particular fax gateway scenario whereby it would have to translate using the read translation path on a channel the frame being translated would be consumed. When the frame is in the write path it is not permitted to free the frame as the caller expects it to continue to exist. This change makes it so that the frame is only consumed on the read path where it is acceptable to free it. ASTERISK-28900 Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d	5 years ago
Alexander Traud	0a4dffe6f8	pjproject_bundled: Honor --without-pjproject. The previous change missed that 'make' uses 'PJPROJECT_BUNDLED' anyway. ASTERISK-28929 Change-Id: I7ef0e78a06ea391b59d95b99d46bbed3fec4fed9	5 years ago
Pirmin Walthert	e8c6e9ae5d	res_pjsip_logger: use the correct pointer when logging tx_messages to pcap When writing tx messages to pcap files, Asterisk is using the wrong pointer resulting in lots of wasted space. This patch fixes it to use the correct pointer. ASTERISK-28932 #close Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23	5 years ago
sungtae kim	25ae412f75	bridge.c: Fixed null pointer exception If the bridge show all command could not get the bridge snapshot, it causes null pointer exception. Fixed it to check the snapshot is null. ASTERISK-28920 Change-Id: I3521fc1b832bfc69644d0833f2c78177e1e51f58	5 years ago
George Joseph	ca3c22c5f1	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	5 years ago
Pirmin Walthert	c16937cdbe	res_pjsip_logger.c: correct the return value checks when writing to pcap files fwrite() does return the number of elements written and not the number of bytes. However asterisk is currently comparing the return value to the size of the written element what means that asterisk logs five WARNING messages on every packet written to the pcap file. This patch changes the code to check for the correct value, which will always be 1. ASTERISK-28921 #close Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2	5 years ago
Joshua C. Colp	9c2871edf4	res_pjsip: Use correct pool for storing the contact_user value. When replacing the user portion of the Contact URI the code was using the ephemeral pool instead of the tdata pool. This could cause the Contact user value to become invalid after a period of time. The code will now use the tdata pool which persists for the lifetime of the message instead. ASTERISK-28794 Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5	5 years ago
Pirmin Walthert	1399f8b4fe	res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header While asterisk is filtering out the x-ast-orig-host parameter from the contact on response messages, it is not filtering it out from the request URI and the to header on SIP requests (for example INVITE). ASTERISK-28884 #close Change-Id: Id032b33098a1befea9b243ca994184baecccc59e	5 years ago
Joshua C. Colp	afa2c9a868	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	5 years ago
Joshua C. Colp	ec7890d7c6	res_sorcery_config: Always reload configuration on errors. When a configuration file in Asterisk is loaded information about it is stored such that on a reload it is not reloaded if nothing has changed. This can be problematic when an error exists in a configuration file in PJSIP since the error will be output at start and not subsequently on reload if the file is unchanged. This change makes it so that if an error is encountered when res_sorcery_config is loading a configuration file a reload will always read in the configuration file, allowing the error to be seen easier. Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed	5 years ago

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