207 Commits (8ad4cb901b97fa077a9420bb9c9901512101c01e)

Author	SHA1	Message	Date
George Joseph	e4de46e847	bridge_softmix: Suppress error on topology change failure ... There are conditions under which a failure to change topology is expected so there's no need to print an ERROR message. ASTERISK-29618 Reported by: Alexander Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481	4 years ago
George Joseph	3b0a53f257	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
George Joseph	8d9633074e	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	5 years ago
George Joseph	6abf6f345d	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	65088494cb	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	5 years ago
Joshua C. Colp	ef608dec78	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	6 years ago
Joshua C. Colp	420d38555e	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
Joshua C. Colp	48a3182c42	bridge_softmix: Always remove audio from mixed frame. ... When receiving audio from a channel we determine if it is talking or silence based on a threshold value. If this threshold is met we always mix the audio into the conference bridge. If this threshold is not met we also mix the audio into the conference bridge UNLESS the drop silence option is enabled. The code that removed the audio from the mixed frame assumed that it was always not present if it did not meet the threshold to be considered talking. This is incorrect. If it has been stated that the audio was mixed into the mixed frame then it has been mixed into the mixed frame. By not removing audio that was considered non-talking it was possible for a channel to receive a slight echo of audio of itself at times. This change ensures that the audio is always removed from the mixed frame going back to the channel so it no longer receives the slight echo. ASTERISK-28898 Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb	6 years ago
Joshua C. Colp	2128eb1f47	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Joshua C. Colp	423b0e68ce	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	6 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Frederic LE FOLL	31173f6586	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. ... During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Kevin Harwell	3891a953cf	bridge_softmix: clear hold when joining a softmix bridge ... MOH continues to play to a channel if that channel was on hold prior to entering a softmix bridge. MOH will not stop even if the original "holder" attempts an unhold. For the most part a softmix bridge ignores holds, so a participating channel shouldn't join while on hold. This patch checks to see if the channel joining the softmix bridge is currently on hold. If so then it indicates an unhold. ASTERISK-28618 Change-Id: I66ccd4efc80f5b4c3dd68186b379eb442916392b	6 years ago
Joshua Colp	d861ebdca8	app_confbridge: Add "all" variants of REMB behavior. ... When producing a combined REMB value the normal behavior is to have a REMB value which is unique for each sender based on all of their receivers. This can result in one sender having low bitrate while all the rest are high. This change adds "all" variants which produces a bridge level REMB value instead. All REMB reports are combined together into a single REMB value that is the same for each sender. ASTERISK-28401 Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c	7 years ago
Kevin Harwell	94eeba6147	bridge_softmix: use a float type to store the internal REMB bitrate ... REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using an unsigned integer to represent the bitrate. However, that type is not large enough to hold all potential bitrate values. If the bitrate is large enough bits were being shifted off the "front" of the mantissa, which caused the wrong value to be sent to the browser. This patch makes it so it now uses a float type to hold the bitrate. Using a float allows for all bitrate values to be correctly represented. ASTERISK-28255 Change-Id: Ice00fdd16693b16b41230664be5d9f0e465b239e	7 years ago
Xiemin Chen	f6cf837aed	bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix ... To avoid the stream name collide if there're more than one video track in one client. If client has multi video tracks, the name of ast_stream which represents each video track may be the same. Use the MSID:LABEL here because it's identifiable. ASTERISK-28196 #close Reported-by: xiemchen Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b	7 years ago
Sean Bright	640aac768b	bridge_builtin_features.c: Set auto(mix)mon variables on both channels ... This is how features behaved up through Asterisk 11, but was changed when the new bridging framework was implemented in Asterisk 12. Reported by rrittgarn in #asterisk. Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b	7 years ago
Robert Cripps	df703c1853	bridge_native_rtp.c: Fail native bridge if no framing match. ... ASTERISK-28110 #close Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607	7 years ago
George Joseph	fec66b8f01	bridge_softmix: Add SDP "label" attribute to streams ... Adding the "label" attribute used for participant info correlation was previously done in app_confbridge but it wasn't working correctly because it didn't have knowledge about which video streams belonged to which channel. Only bridge_softmix has that data so now it's set when the bridge topology is changed. ASTERISK-28107 Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499	7 years ago
Alexander Traud	804d931f27	bridge_softmix_binaural: Enable FFTW3 in Solaris 11. ... ASTERISK-27939 Change-Id: Ice5640e08385a64a0a6555deaccd91e86bca154f	8 years ago
George Joseph	017b7849bc	Merge "VECTOR: Passing parameters with side effects to macros is dangerous."	8 years ago
Richard Mudgett	cad50d6dbf	VECTOR: Passing parameters with side effects to macros is dangerous. ... * Fix several instances where we were bumping a ref in the parameter and then unrefing the object if it failed. The way the AST_VECTOR_APPEND() and AST_VECTOR_REPLACE() macros are implemented means if it fails the new value was never evaluated. Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a	8 years ago
Richard Mudgett	aaaa6f4a4b	bridge_softmix.c: Fix memory leak. ... Made release the media_types vector in softmix_bridge_stream_topology_changed(). Change-Id: Ide3f47e379b614879220dfeeb13843f9f2b177b5	8 years ago
Joshua Colp	7528b86cad	stream: Make the topology a reference counted object. ... The stream topology has no lock of its own resulting in another lock protecting it in some way (for example the channel lock). If multiple channels are being juggled at the same time this can be problematic. This change makes the topology a reference counted object instead which guarantees it will remain valid even without the channel lock being held. Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03	8 years ago
Jenkins2	f633af89c1	Merge "bridge_softmix: Fix sporadic incorrect video stream mapping."	8 years ago
George Joseph	3bad41257b	Merge "BuildSystem: Add DragonFly BSD."	8 years ago
Jenkins2	9c430569d4	Merge "bridge_softmix: Forward TEXT frames"	8 years ago
Joshua Colp	de9c0ede4a	bridge_softmix: Fix sporadic incorrect video stream mapping. ... When an externally initiated renegotiation occurred it was possible for video streams to be incorrectly remapped, resulting in no video flowing to some receivers. This change ensures that only the video source sets up mappings and also that removed streams do not have mappings set up. Change-Id: Iab05f2254df3606670774844bb0935f833d3a9b0	8 years ago
Joshua Colp	5712a0ae52	bridge_softmix: Fix some REMB bugs. ... This change fixes a bug where a REMB collector may be freed twice, and also tweaks REMB combining such that if there is no bitrate from anyone (or there are no sources) we report 0 instead of using an old bitrate. ASTERISK-27804 Change-Id: Ia9dc9c150043890ee7ff85e9cdec007f1a77fcfd	8 years ago
Alexander Traud	efe40ff671	BuildSystem: Add DragonFly BSD. ... ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	8 years ago
Joshua Colp	8de3fa2b56	bridge_softmix / app_confbridge: Add support for REMB combining. ... This change adds the ability for multiple REMB reports in bridge_softmix to be combined according to a configured behavior into a single report. This single report is sent back to the sender of video, which adjusts the encoding bitrate to be at or below the bitrate of the report. The available behaviors are: lowest, highest, and average. Lowest uses the lowest received bitrate. Highest uses the highest received bitrate. Average goes through the received bitrates adding them to the previous average and creates a new average. Other behaviors can be added in the future and the existing average one may be adjusted, but this provides the foundation to do so. Support for configuring which behavior to use has been added to app_confbridge. ASTERISK-27804 Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66	8 years ago
George Joseph	4fb7967c73	bridge_softmix: Forward TEXT frames ... Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
Kevin Harwell	cf73a4203f	bridge_softmix: Clear "talking" when a channel is put on hold ... This patch clears the talking flag from the channel (if already set), and notifies listeners when that channel is put on hold. Note however, if the endpoint continues to send audio frames and these are received by the bridge then that channel will be put back into a "talking" state even though they are on hold. ASTERISK-27755 #close Change-Id: I930e16c4662810f9f02043d69062f88173c5e2ef	8 years ago
Richard Mudgett	39f733406d	bridge_simple.c: Fix stream topology handling. ... The handling of stream topologies was not protected by channel locks in simple_bridge_request_stream_topology_change(). * Fixed topology handling to be protected by channel locks where needed in simple_bridge_request_stream_topology_change(). ASTERISK-27692 Change-Id: Ica5d78a6c7ecf4f0b95fb16de28d3889b32c4776	8 years ago
Joshua Colp	dd459da428	Merge "bridge_softmix.c: Report not talking immediately when muted."	8 years ago
Richard Mudgett	1ccac0be0e	bridge_softmix.c: Report not talking immediately when muted. ... Currently in app_confbridge if someone mutes a channel while that channel is talking, the talk detection code is suspended while the channel is muted. As far an an external observer is concerned, the muted channel's talk status is still "talking" even though the channel is not contributing audio to the conference bridge. When the channel is later unmuted, it takes the usual 'dsp_silence_threshold' option time to clear the talking status even though the channel may have stopped talking while the channel was muted. * In bridge_softmix.c, clear the talking status and report talking stopped if the channel was talking when the channel is muted. When the channel is unmuted and the channel is still talking then report the channel as talking since it is contributing audio to the bridge again. ASTERISK-27647 Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e	8 years ago
Richard Mudgett	b9024197ab	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. ... The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	8 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	8 years ago
Corey Farrell	d46cbe788a	bridge_softmix: Fix sfu_append_source_streams test. ... * validate_stream: zero result from ast_format_cap_identical indicates they are not identical, rather than non-zero indicating an error. * validate_original_streams: use num_streams instead of ARRAY_LEN(params). * Fix declaration of alice_dest_stream and bob_dest_stream. Change-Id: I6b1dd8bed10439d3c7406f033eb1896b6c419147	8 years ago
Alexander Traud	b12c8cffad	bridge_softmix: Removed unused parameter from check_binaural_position_change(.). ... Found as a result of the function being passed an uninitalized variable by clang. ASTERISK-27550 Change-Id: I8af3bd84656b685a956d498459f8db3613f68954	8 years ago
Richard Mudgett	fa36f9c01b	bridge_native_rtp.c: Fix reentrancy framehook crash. ... If two channels enter different native rtp bridges at the same time it is possible that the framehook interface data pointer can be corrupted because the struct variable was declared static. * Fixed the reentrancy corruption by changing the framehook interface struct static variable to a stack local variable. * Moved the hook.data assignment outside of the channel lock. It did not need the lock's protection. It probably was giving a false sense of security. The testsuite channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up test caught this with MALLOC_DEBUG and DO_CRASH enabled. Change-Id: If9e35b97d19209b0f984941c1d8eb5f7c55eea91	8 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Jenkins2	7384abc5c1	Merge "bridge_softmix.c: Don't match dead streams."	8 years ago
Jenkins2	ece673f0f6	Merge "bridge_softmix.c: Change remove_destination_streams() return meaning."	8 years ago
Richard Mudgett	4c04e13783	bridge_softmix.c: Change remove_destination_streams() return meaning. ... The return value of remove_destination_streams() now means we removed a stream from the topology by making it a dead stream. Now we won't try to request a topology change if we didn't remove any streams. Change-Id: Icd91571d856a1d04299a24c411e325c1d9d5c61d	8 years ago
Richard Mudgett	ea4179599f	bridge_softmix.c: Don't match dead streams. ... * Made is_video_source() and is_video_dest() not match dead streams. * Optimized is_video_dest() to reduce duplicated code. Change-Id: I4e7ab762c7ee98395e78e6516399f57a2609b9a1	8 years ago
Corey Farrell	91d9eae79b	bridge_softmix: Fix memory leaks. ... Change-Id: Ifaf3e93b398595d21d07f535330fef77ff15a80c	8 years ago
Corey Farrell	05f557820b	bridge_softmix: Note why ast_stream_topology_set_stream cannot fail. ... This appeared in my audit of ast_stream_topology_set_stream callers not checking for errors but in this situation the call cannot fail. Add comment so this can be ignored in the future. Change-Id: I91d25704859efbe50b8b82cfe1cd3c40ba177c9f	8 years ago
Richard Mudgett	ee08f10d06	Fix ast_(v)asprintf() malloc failure usage conditions. ... When (v)asprintf() fails, the state of the allocated buffer is undefined. The library had better not leave an allocated buffer as a result or no one will know to free it. The most likely way it can return failure is for an allocation failure. If the printf conversion fails then you actually have a threading problem which is much worse because another thread modified the parameter values. * Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL on failure. That is much more useful than either an uninitialized pointer or a pointer that has already been freed. Many uses won't have to check for failure to ensure that the buffer won't be double freed or prevent an attempt to free an uninitialized pointer. * stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by ast_asprintf(). * ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to the wrong thing which is now not needed even if assigning to the right thing. Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23	8 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	8 years ago