797 Commits (a80ae57cac1ee4ccab0a44c6cb45d1c184e5ca41)

Author	SHA1	Message	Date
Peter Jannesen	f3e74d34ce	channel: Preserve CHANNEL(userfield) on masquerade. ... In certain circumstances a channel may undergo an operation referred to as a masquerade. If this occurs the CHANNEL(userfield) value was not preserved causing it to get lost. This change makes it so that this field is now preserved. Fixes: #882	8 months ago
Ben Ford	3841fa814e	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	9 months ago
Naveen Albert	051ddbe729	file.c, channel.c: Don't emit warnings if progress received. ... Silently ignore AST_CONTROL_PROGRESS where appropriate, as most control frames already are. Resolves: #696	1 year ago
George Joseph	a5ae546b88	Reduce startup/shutdown verbose logging ... When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582	1 year ago
Naveen Albert	67088b256d	func_frame_trace: Add CLI command to dump frame queue. ... This adds a simple CLI command that can be used for analyzing all frames currently queued to a channel. A couple log messages are also adjusted to be more useful in tracing bridging problems. Resolves: #533	1 year ago
George Joseph	c31cd32b82	Revert "core & res_pjsip: Improve topology change handling." ... This reverts commit 315eb551db. Over the past year, we've had several reports of "topology storms" occurring where 2 external facing channels connected by one or more local channels and bridges will get themselves in a state where they continually send each other topology change requests. This usually manifests itself in no-audio calls and a flood of "Exceptionally long queue length" messages. It appears that this commit is the cause so we're reverting it for now until we can determine a more appropriate solution. Resolves: #530	1 year ago
Holger Hans Peter Freyther	9fd2655d5a	stasis: Update the snapshot after setting the redirect ... The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest.	2 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ... ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	2 years ago
Maximilian Fridrich	315eb551db	core & res_pjsip: Improve topology change handling. ... This PR contains two relatively separate changes in channel.c and res_pjsip_session.c which ensure that topology changes are not ignored in cases where they should be handled. For channel.c: The function ast_channel_request_stream_topology_change only triggers a stream topology request change indication, if the channel's topology does not equal the requested topology. However, a channel could be in a state where it is currently "negotiating" a new topology but hasn't updated it yet, so the topology request change would be lost. Channels need to be able to handle such situations internally and stream topology requests should therefore always be passed on. In the case of chan_pjsip for example, it queues a session refresh (re-INVITE) if it is currently in the middle of a transaction or has pending requests (among other reasons). Now, ast_channel_request_stream_topology_change always indicates a stream topology request change even if the requested topology equals the channel's topology. For res_pjsip_session.c: The function resolve_refresh_media_states does not process stream state changes if the delayed active state differs from the current active state. I.e. if the currently active stream state has changed between the time the sip session refresh request was queued and the time it is being processed, the session refresh is ignored. However, res_pjsip_session contains logic that ensures that session refreshes are queued and re-queued correctly if a session refresh is currently not possible. So this check is not necessary and led to some session refreshes being lost. Now, a session refresh is done even if the delayed active state differs from the current active state and it is checked whether the delayed pending state differs from the current active - because that means a refresh is necessary. Further, the unit test of resolve_refresh_media_states was adapted to reflect the new behavior. I.e. the changes to delayed pending are prioritized over the changes to current active because we want to preserve the original intention of the pending state. ASTERISK-30184 Change-Id: Icd0703295271089057717006730b555b9a1d4e5a	3 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Naveen Albert	3a3b8fbd9f	channel.c: Clean up debug level 1. ... Although there are 10 debugs levels, over time, many current debug calls have come to use inappropriately low debug levels. In particular, a select few debug calls (currently all debug 1) can result in thousands of debug messages per minute for a single call. This can adds a lot of noise to core debug which dilutes the value in having different debug levels in the first place, as these log messages are from the core internals are are better suited for higher debug levels. Some debugs levels are thus adjusted so that debug level 1 is not inappropriately overloaded with these extremely high-volume and general debug messages. ASTERISK-29897 #close Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4	3 years ago
Naveen Albert	68f1e5d508	ami: Add AMI event for Wink ... Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	3 years ago
Naveen Albert	70bc0ff9d0	documentation: Add missing AMI documentation ... Adds missing documentation for some channel, bridge, and queue events. ASTERISK-24427 ASTERISK-29515 Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64	3 years ago
Naveen Albert	ee9eef492c	app_mf: Add full tech-agnostic MF support ... Adds tech-agnostic support for MF signaling by adding MF sender and receiver applications as well as Dial integration. ASTERISK-29496-mf #do-not-close Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4	3 years ago
Sean Bright	e9cac5f4bf	channel: Short-circuit ast_channel_get_by_name() on empty arg. ... We know that passing a NULL or empty argument to ast_channel_get_by_name() will never result in a matching channel and will always result in an error being emitted, so just short-circuit out in that case. ASTERISK-28219 #close Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce	4 years ago
Alexander Traud	fa91010229	channel: Fix for Doxygen. ... ASTERISK-29751 Change-Id: Ie04da5029c57ebee44733bdf05013156abe80176	4 years ago
Josh Soref	f382775241	main: Spelling fixes ... Correct typos of the following word families: analysis nuisance converting although transaction desctitle acquire update evaluate thousand this dissolved management integrity reconstructed decrement further on irrelevant currently constancy anyway unconstrained featuregroups right larger evaluated encumbered languages digits authoritative framing blindxfer tolerate traverser exclamation perform permissions rearrangement performing processing declension happily duplicate compound hundred returns elicit allocate actually paths inheritance atxferdropcall earlier synchronization multiplier acknowledge across against thousands joyous manipulators guaranteed emulating soundfile ASTERISK-29714 Change-Id: I926ba4b11e9f6dd3fdd93170ab1f9b997910be70	4 years ago
Sean Bright	ce2d743d59	various: Fix GCC 11.2 compilation issues. ... * Initialize some variables that are never used anyway. * Use valid pointers instead of integers cast to void pointers when calling pthread_setspecific(). ASTERISK-29711 #close ASTERISK-29713 #close Change-Id: I8728cd6f2f4b28e0e48113c5da450b768c2a6683	4 years ago
Naveen Albert	ddf6299b8d	func_sayfiles: Retrieve say file names ... Up until now, all of the logic used to translate arguments to the Say applications has been directly coupled to playback, preventing other modules from using this logic. This refactors code in say.c and adds a SAYFILES function that can be used to retrieve the file names that would be played. These can then be used in other applications or for other purposes. Additionally, a SayMoney application and a SayOrdinal application are added. Both SayOrdinal and SayNumber are also expanded to support integers greater than one billion. ASTERISK-29531 Change-Id: If9718c89353b8e153d84add3cc4637b79585db19	4 years ago
Joshua C. Colp	5382b9dbb8	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	4 years ago
Naveen Albert	04454fc238	AMI: Add AMI event to expose hook flash events ... Although Asterisk can receive and propogate flash events, it currently provides no mechanism for doing anything with them itself. This AMI event allows flash events to be processed by Asterisk. Additionally, AST_CONTROL_FLASH is included in a switch statement in channel.c to avoid throwing a warning when we shouldn't. ASTERISK-29380 Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81	4 years ago
Joshua C. Colp	cc127a999c	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	4 years ago
Joshua C. Colp	3e5b9e3952	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	4 years ago
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Alexander Traud	6d980de282	channel: Set up calls without audio (text+video), again. ... ASTERISK-29259 Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5	4 years ago
Jasper van der Neut	e831952eba	channels: Don't dereference NULL pointer ... Check result of ast_translator_build_path against NULL before dereferencing. ASTERISK-29091 Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29	5 years ago
George Joseph	44bb0858cb	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	5 years ago
George Joseph	647c53c41f	ACN: Changes specific to the core ... Allow passing a topology from the called channel back to the calling channel. * Added a new function ast_queue_answer() that accepts a stream topology and queues an ANSWER CONTROL frame with it as the data. This allows the called channel to indicate its resolved topology. * Added a new virtual function to the channel tech structure answer_with_stream_topology() that allows the calling channel to receive the called channel's topology. Added ast_raw_answer_with_stream_topology() that invokes that virtual function. * Modified app_dial.c and features.c to grab the topology from the ANSWER frame queued by the answering channel and send it to the calling channel with ast_raw_answer_with_stream_topology(). * Modified frame.c to automatically cleanup the reference to the topology on ANSWER frames. Added a few debugging messages to stream.c. Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c	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
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
Nathan Bruning	f217fcdc62	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions ... Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	5 years ago
Kevin Harwell	3c345ec56d	channel: write to a stream on multi-frame writes ... If a frame handling routine returns a list of frames (vs. a single frame) those frames are never passed to a tech's write_stream handler even if one is available. For instance, if a codec translation occurred and that codec returned multiple frames then those particular frames were always only sent to the tech's "write" handler. If that tech (pjsip for example) was stream capable then those frames were essentially ignored. Thus resulting in bad audio. This patch makes it so the "write_stream" handler is appropriately called for all cases, and for all frames if available. ASTERISK-28795 #close Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d	5 years ago
Joshua C. Colp	5a5be92b79	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	5 years ago
George Joseph	fd823225a6	channel.c: Resolve issue with receiving SIP INFO packets for DTMF ... The problem is essentially the same as in ASTERISK~28245. Besides the direct media scenario we have an additional scenario where a special client is involved. This device mutes audio by default in transmit direction (no rtp frames) and activates audio only by a foot switch. In this situation dtmf input (pin for conferences, transfer features codes , etc) using SIP INFO mode is not understood properly especially when SIP INFO messages are sent quickly. This patch ensures that SIP INFO frames are properly queued and processed in the above scenario. The patch also corrects situations where successive dtmf events are received quicker than the signalled event duration (plus minimum gap/pause) allows, i.e. DTMF events have to be buffered in the ast channel read queue and emulation has to be processed asynchronously at slower speed. Reported by: Thomas Arimont patches: trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525) Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2	6 years ago
Kevin Harwell	3656c42cb0	various modules: json integer overflow ... There were still a few places in the code that could overflow when "packing" a json object with a value outside the base type integer's range. For instance: unsigned int value = INT_MAX + 1 ast_json_pack("{s: i}", value); would result in a negative number being "packed". In those situations this patch alters those values to a ast_json_int_t, which widens the value up to a long or long long. ASTERISK-28480 Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1	6 years ago
Antoni Goldstein	8e21c25ce5	app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings ... Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled at the earliest received PROGRESS or RINGING. Added millisecond versions of DIALEDTIME and ANSWEREDTIME. Added millisecond versions of ast_channel_get_up_time and ast_channel_get_duration in channel.c. ASTERISK-28363 Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1	6 years ago
Valentin Vidic	17f76d27cc	channel.c: Fix segfault with Monitor(wav,file,i) ... If the Monitor is started with the i option the read_stream will be NULL. One code path in channel.c checks if write_stream is set but than uses read_stream instead causing a segfault. ASTERISK-28249 Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525	6 years ago
mohitdhiman	d60ee2eeae	stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure. ... During Bridging of two channels if masquerade operation is performed on a channel (clone channel) which was created with endpoint details (ast_channel_alloc_with_endpoint()) and the original channel which is created without endpoint details (ast_channel_alloc()) then both the channels must exchange their endpoint details or else after masquerade when clone channel is being destroyed the endpoint cleanup callbacks will be destroyed too and after call completion unique_id of original channel will still be there in ast_endpoint structure's channel_ids container. ASTERISK-28197 Change-Id: I97ce73da390af20fd082fb09d722a6fe9cb2f39d	6 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago
Joshua Colp	d0ccbb3377	stasis: Use an implementation specific channel snapshot cache. ... Channels no longer use the Stasis cache for channel snapshots. Instead they are stored in a hash table in stasis_channels which reduces the number of Stasis messages created and allows better storage. As a result the following APIs are no longer available since the stasis cache is no longer used: ast_channel_topic_cached() ast_channel_topic_all_cached() The ast_channel_cache_all() and ast_channel_cache_by_name() functions now return an ao2_container of ast_channel_snapshots rather than a container of stasis_messages therefore you can't (and don't need to) call stasis_cache functions on it. The ast_channel_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. The ast_channel_snapshot_type() stasis message now has the ast_channel_snapshot_update structure as it's data. It contains the last snapshot and the new one. ast_channel_snapshot_get_latest() still returns the latest snapshot. The latest snapshot is now stored on the channel itself to eliminate cache hits when Stasis messages that have the snapshot as a payload are created. ASTERISK-28102 Change-Id: I9334febff60a82d7c39703e49059fa3a68825786	7 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Corey Farrell	687ab7aeee	astobj2: Eliminate legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
neutrino88	17f4e6ad4d	core/frame: generate correct T.140 payload in ast_sendtext_data() ... ast_sendtext_data() would create an incorrect T.140 text frame which length include the null terminator byte. It causes ultimately RTP packets to be send with this trailing 0. The proposed fix just set the correct length to the text frame ASTERISK-28089 Reported by: Emmanuel BUU Tested by: Emmanuel BUU Change-Id: I7ab1b9ed1e21683b2b667ea0a59d9aba3c77dd96	7 years ago
George Joseph	4d51a8e05b	channel.c: Address stack overflow in does_id_conflict() ... does_id_conflict() was passing a pointer to an int to a callback that expected a pointer to a size_t. Found by the Address Sanitizer. Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503	7 years ago
Joshua Colp	f97d92bd0a	core: Don't stop generators when writing RTCP frames. ... Generators provide such functionality as tone generation or silence generation. RTCP frames provide RTCP information and should not stop generators from operating. ASTERISK-28005 Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9	7 years ago
Richard Mudgett	7a238fe74d	AMI SendText action: Fix to use correct thread to send the text. ... The AMI action was directly sending the text to the channel driver. However, this makes two threads attempt to handle media and runs afowl of CHECK_BLOCKING. * Queue a read action to make the channel's media handling thread actually send the text message. This changes the AMI actions success/fail response to just mean the text was queued to be sent not that the text actually got sent. The channel driver may not even support sending text messages. ASTERISK-27943 Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379	7 years ago
George Joseph	e3585353f6	res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs ... In addition to text/* content types, incoming_in_dialog_request now accepts application/* content types. Also fixed a length issue when copying the body text. It was one character short. ASTERISK-27942 Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818	7 years ago
George Joseph	d87631d21f	Merge changes from topic 'ASTERISK-27625' ... * changes: channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. channel.c: Fix usage of CHECK_BLOCKING() autoservice: Don't start channel autoservice if the thread is a user interface.	7 years ago