mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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709 Commits (5d938045d465d66e28bb49816c482a8b68f99342)
Author | SHA1 | Message | Date |
---|---|---|---|
|
5d938045d4 |
channel: Remove old epoll support and fixed max number of file descriptors.
This change removes the old epoll support which has not been used or maintained in quite some time. The fixed number of file descriptors on a channel has also been removed. File descriptors are now contained in a growable vector. This can be used like before by specifying a specific position to store a file descriptor at or using a new API call, ast_channel_fd_add, which adds a file descriptor to the channel and returns its position. Tests have been added which cover the growing behavior of the vector and the new API call. ASTERISK-26885 Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928 |
8 years ago |
|
3ed05badb9 |
core: Add stream topology changing primitives with tests.
This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9 |
8 years ago |
|
0560c32375 |
stream: Unit tests for stream read and tweaks framework
* Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2 |
8 years ago |
|
c07c6714f2 |
channel: Add ast_read_stream function for reading frames from all streams.
This change introduces an ast_read_stream function and callback in the channel technology which allows reading frames from all streams and not just the default streams. The stream number has also been added to frames. This is to allow the case where frames are queued onto the channel instead of being read directly from the driver. This change does impose a restriction on reading though: a chain of frames can only contain frames from the same stream. ASTERISK-26816 Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce |
8 years ago |
|
6cc890b880 |
channel: Add support for writing to a specific stream.
This change adds an ast_write_stream function which allows writing a frame to a specific media stream. It also moves ast_write() to using this underneath by writing media frames provided to it to the default streams of the channel. Existing functionality (such as audiohooks, framehooks, etc) are limited to being applied to the default stream only. Unit tests have also been added which test the behavior of both non-multistream and multistream channels to confirm that the write() and write_stream() callbacks are invoked appropriately. ASTERISK-26793 Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c |
8 years ago |
|
ab34e46b3a |
Merge "stream: Rename creates/destroys to allocs/frees"
|
8 years ago |
|
f8f513d363 |
stream: Rename creates/destroys to allocs/frees
To be consistent with sdp implementation. Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500 |
8 years ago |
|
ca7fa7bbd2 |
Merge "stream: Add stream topology to channel"
|
8 years ago |
|
bf2f091bbb |
stream: Add stream topology to channel
Adds topology set and get to channel. ASTERISK-26790 Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4 |
8 years ago |
|
cea835e565 |
Merge "channel: Protect flags in ast_waitfor_nandfds operation."
|
8 years ago |
|
89871576b9 |
channel: Protect flags in ast_waitfor_nandfds operation.
The ast_waitfor_nandfds operation will manipulate the flags of channels passed in. This was previously done without the channel lock being held. This could result in incorrect values existing for the flags if another thread manipulated the flags at the same time. This change locks the channel during flag manipulation. ASTERISK-26788 Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed |
8 years ago |
|
16fdb11bc3 |
core: Cleanup some channel snapshot staging anomalies.
We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a |
8 years ago |
|
50029f585e |
channel.c: Fix unbalanced read queue deadlocking local channels.
Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 |
8 years ago |
|
97c308471d |
res_agi: Prevent an AGI from eating frames it should not. (Re-do)
A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 |
8 years ago |
|
72e3fc5845 |
Frame deferral: Revert API refactoring.
There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 |
8 years ago |
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6f645a6d4e |
Merge "media: Add experimental support for RTCP feedback."
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8 years ago |
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20aed30d9a |
T.140: Fix format ref and memory leaks.
* channel.c:ast_sendtext(): Fix T.140 SendText memory leak. * format_compatibility.c: T.140 RED and T.140 were swapped. * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic scheduled red_write(). * res_rtp_asterisk.c: Some other minor misc tweaks. Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb |
8 years ago |
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1061539b75 |
media: Add experimental support for RTCP feedback.
This change adds experimental support for providing RTCP feedback information to codec modules so they can dynamically change themselves based on conditions. ASTERISK-26584 Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857 |
8 years ago |
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283c16c6b6 |
abstract/fixed/adpative jitter buffer: disallow frame re-inserts
It was possible for a frame to be re-inserted into a jitter buffer after it had been removed from it. A case when this happened was if a frame was read out of the jitterbuffer, passed to the translation core, and then multiple frames were returned from said translation core. Upon multiple frames being returned the first is passed on, but sebsequently "chained" frames are put back into the read queue. Thus it was possible for a frame to go back into the jitter buffer where this would cause problems. This patch adds a flag to frames that are inserted into the channel's read queue after translation. The abstract jitter buffer code then checks for this flag and ignores any frames marked as such. Change-Id: I276c44edc9dcff61e606242f71274265c7779587 |
8 years ago |
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621d886ca7 |
Frame deferral: Re-queue deferred frames one-at-a-time.
The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d |
9 years ago |
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e5e887be53 |
chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no
The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d |
9 years ago |
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97b2ba472d |
Merge "channel: Fix issues in hangup scenarios caused by frame deferral"
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9 years ago |
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b8e91bb9cc |
Merge "Revert "Revert "channel: Use frame deferral API for safe sleep."""
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9 years ago |
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013e7dd4a6 |
Merge "Revert "Revert "AGI: Only defer frames when in an interception routine."""
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9 years ago |
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ac0a1ee6da |
Merge "Revert "Revert "Add API for channel frame deferral."""
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9 years ago |
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7263a17ca0 |
channel: Fix issues in hangup scenarios caused by frame deferral
ASTERISK-26343
Change-Id: I06dbf7366e26028251964143454a77d017bb61c8
(cherry picked from commit
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9 years ago |
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0dc4567133 |
Revert "Revert "channel: Use frame deferral API for safe sleep.""
This reverts commit
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9 years ago |
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f62c9c42fa |
Revert "Revert "AGI: Only defer frames when in an interception routine.""
This reverts commit
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9 years ago |
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2966fa5ad7 |
Revert "Revert "Add API for channel frame deferral.""
This reverts commit
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9 years ago |
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c6d755de11 |
res_ari: Add support for channel variables in ARI events.
This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229 |
9 years ago |
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fa749866c1 |
Revert "Add API for channel frame deferral."
This reverts commit
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9 years ago |
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f073f648b8 |
Add API for channel frame deferral.
There are several places in Asterisk that have duplicated logic for deferring important frames until later. This commit adds a couple of API calls to facilitate this automatically. ast_channel_start_defer_frames(): Future reads of deferrable frames on this channel will be deferred until later. ast_channel_stop_defer_frames(): Any frames that have been deferred get requeued onto the channel. ASTERISK-26343 Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641 |
9 years ago |
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a6e5bae3ef |
Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 |
9 years ago |
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e459b8dadf |
ARI: Detect duplicate channel IDs
ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06 |
9 years ago |
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8b58db0962 |
Merge "Binaural synthesis (confbridge): interleaved two-channel audio."
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9 years ago |
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7af7490e42 |
audiohooks: Remove redundant codec translations when using audiohooks
The main frame read and write handlers in main/channel.c don't use the optimum placement in the processing flow for calling audiohooks callbacks, as far as codec translation is concerned. This change places the audiohooks callback code: * After the channel read translation if the frame is not linear before the translation, thereby increasing the chance that the frame is linear as required by audiohooks * Before the channel write translation if the frame is linear at this point This prevents the audiohooks code from instantiating additional translation paths to/from linear where a linear frame format is already available, saving valuable CPU cycles ASTERISK-26419 Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f |
9 years ago |
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c455823657 |
Binaural synthesis (confbridge): interleaved two-channel audio.
Asterisk only supports mono audio at the moment. This patch adds interleaved two-channel audio to Asterisk's channels. ASTERISK-26292 Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a |
9 years ago |
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858fa5eb2c |
channel: No hung-up on failing security requirements.
In your Diaplan, if you specify same => n,Set(CHANNEL(secure_bridge_media)=1) same => n,Set(CHANNEL(secure_bridge_signaling)=1) only the SIP channel driver chan_sip supports this. All other channels drivers like res_pjsip fail. In case of failure, the original sRTP source code released the whole channel, even if not hung-up, yet. This change does not release the channel but instead hangs-up the channel. ASTERISK-26306 Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db |
9 years ago |
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29b0f733a0 |
Add missing checks during startup.
This ensures startup is canceled due to allocation failures from the following initializations. * channel.c: ast_channels_init * config_options.c: aco_init ASTERISK-26265 #close Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611 |
9 years ago |
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e739888d99 |
res_pjsip: Add fax_detect_timeout endpoint option.
The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d |
9 years ago |
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3e8d523d88 |
core/dial: New channel variable FORWARDERNAME
Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 |
9 years ago |
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c7d45b84f9 |
bridge core: Add owed T.38 terminate when channel leaves a bridge.
The channel is now going to get T.38 terminated when it leaves the bridging system and the bridged peers are going to get T.38 terminated as well. ASTERISK-25582 Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7 |
9 years ago |
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128c96456c |
channel.c: Route all control frames to a channel through the same code.
Frame hooks can conceivably return a control frame in exchange for an audio frame inside ast_write(). Those returned control frames were not handled quite the same as if they were sent to ast_indicate(). Now it doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a channel or ast_indicate(). ASTERISK-25582 Change-Id: I5775f41421aca2b510128198e9b827bf9169629b |
9 years ago |
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fa20729032 |
Audit improper usage of scheduler exposed by 5c713fdf18 .
channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8 |
10 years ago |
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cf79b62778 |
ast_format_cap_get_names: To display all formats, the buffer was increased.
ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a |
10 years ago |
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f42084be09 |
Do not swallow frames on channels leaving bridges.
When leaving a bridge, indications on a channel could be swallowed by the internal indication logic because it appears that the channel is on its way to be hung up anyway. One such situation where this is detrimental is when channels on hold are redirected out of a bridge. The AST_CONTROL_UNHOLD indication from the bridging code is swallowed, leaving the channel in question to still appear to be on hold. The fix here is to modify the logic inside ast_indicate_data() to not drop the indication if the channel is simply leaving a bridge. This way, channels on hold redirected out of a bridge revert to their expected "in use" state after the redirection. ASTERISK-25418 #close Reported by Mark Michelson Change-Id: If6115204dfa0551c050974ee138fabd15f978949 |
10 years ago |
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8fb8988fd4 |
Add a test event for inband ringing.
This event is necessary for the bridge_wait_e_options test to be able to confirm that ringing is being played on the local channel that runs the BridgeWait() application with the e(r) option. ASTERISK-25292 #close Reported by Kevin Harwell Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e |
10 years ago |
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687597ca8c |
holding_bridge: ensure moh participants get frames
Currently, if a blank musiconhold.conf is used, musiconhold will fail
to start for a channel going into a holding bridge with an anticipation
of getting music on hold. That being the case, no frames will be written
to the channel and that can pose a problem for blind transfers in PJSIP
which may rely on frames being written to get past the REFER framehook.
This patch makes holding bridges start a silence generator if starting
music on hold fails and makes it so that if no music on hold functions
are installed that the ast_moh_start function will report a failure so
that consumers of that function will be able to respond appropriately.
ASTERISK-25271 #close
Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
(cherry picked from commit
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10 years ago |
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8d6cf667dc |
channel: Remove ignore of answer on non-outgoing channels.
Due to the way that channels can now be moved around inside of Asterisk it is possible for the outgoing flag of a channel to get cleared before it has been answered. This results in the bridge not receiving notification that the outgoing leg has been answered. This most easily exhibits itself with DTMF based blond transfers. Since the answer of the outgoing leg is ignored the other party continues to receive both a locally generated ringing and the media stream of the outgoing leg upon its answer. This results in no media being heard. This change removes the ignore of the answer and allows it to pass through. ASTERISK-25171 #close Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e |
10 years ago |
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c3ec5da156 |
Remove unneeded uses of optional_api providers.
A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478 |
10 years ago |