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Author	SHA1	Message	Date
under	de3f5350de	codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother ... If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps. This makes the audio stream not-playable at the receiver side. Linphone isn't able to play such an audio - lots of disruptions are heard. Also I had complains of bad audio from users which use other types of phones. After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c). Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c). However, this flag is never set in Asterisk-12 and newer. Previously it has been set (see Asterisk-11). ASTERISK-29526 #close Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d	4 years ago
Sean Bright	6428124b06	res_http_media_cache: Cleanup audio format lookup in HTTP requests ... Asterisk first looks at the end of the URL to determine the file extension of the returned audio, which in many cases will not work because the URL may end with a query string or a URL fragment. If that fails, Asterisk then looks at the Content-Type header and then finally parses the URL to get the extension. The order has been changed such that we look at the Content-Type header first, followed by looking for the extension of the parsed URL. We no longer look at the end of the URL, which was error prone. ASTERISK-29527 #close Change-Id: I1e3f83b339ef2b80661704717c23568536511032	4 years ago
Joshua C. Colp	d0f189a5c9	docs: Remove embedded macro in WaitForCond XML documentation. ... Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c	4 years ago
Ben Ford	db7b025532	Update AMI and ARI versions for Asterisk 20. ... Bumped AMI and ARI versions for the next major Asterisk version (20). Change-Id: I2e65794f206d443178ab6895767fb53f04cc3e6a	4 years ago
Kevin Harwell	e8cda4b32c	AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS ... If an SSL socket parent/listener was destroyed during the handshake, depending on timing, it was possible for the handling callback to attempt access of it after the fact thus causing a crash. ASTERISK-29415 #close Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d	4 years ago
Kevin Harwell	1b62831f2c	AST-2021-008 - chan_iax2: remote crash on unsupported media format ... If chan_iax2 received a packet with an unsupported media format, for example vp9, then it would set the frame's format to NULL. This could then result in a crash later when an attempt was made to access the format. This patch makes it so chan_iax2 now ignores/drops frames received with unsupported media format types. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	4 years ago
Joshua C. Colp	ec16d2ecbd	AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. ... If a re-INVITE is received after we have sent a BYE request then it is possible for no channel to be present on the session. If this occurs we allow PJSIP to produce the offer instead. Since the call is being hung up if it produces an incorrect offer it doesn't actually matter. This also ensures that code which produces SDP does not need to handle if a channel is not present. ASTERISK-29381 Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042	4 years ago
Asterisk Development Team	e6ddbe0922	Update CHANGES and UPGRADE.txt for 19.0.0	4 years ago
Andre Barbosa	f4d3f021f9	res_stasis_playback: Check for chan hangup on play_on_channels ... Verify `ast_check_hangup` before looping to the next sound file. If the call is already hangup we just break the cycle. It also ensures that the PlaybackFinished event is sent if the call was hangup. This is also use-full when we are playing a big list of file for a channel that is hangup. Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played. With the patch we just break the playback cycle when the chan is hangup. ASTERISK-29501 #close Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8	4 years ago
Sean Bright	d5bb27a06f	res_http_media_cache.c: Fix merge errors from 18 -> master ... ASTERISK-27871 #close Change-Id: I6624f2d3a57f76a89bb372ef54a124929a0338d7	4 years ago
Sean Bright	237285a9a8	res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup. ... From RFC 8225 Section 5.2.1: The "dest" claim is a JSON object with the claim name of "dest" and MUST have at least one identity claim object. The "dest" claim value is an array containing one or more identity claim JSON objects representing the destination identities of any type (currently "tn" or "uri"). If the "dest" claim value array contains both "tn" and "uri" claim names, the JSON object should list the "tn" array first and the "uri" array second. Within the "tn" and "uri" arrays, the identity strings should be put in lexicographical order, including the scheme-specific portion of the URI characters. Additionally, make it clear that there was a failure to sign the JWT payload and not necessarily a memory allocation failure. Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9	4 years ago
Sean Bright	d568326807	res_http_media_cache.c: Parse media URLs to find extensions. ... Use cURL's URL parsing API, falling back to the urlparser library, to parse playback URLs in order to find their file extensions. For backwards compatibility, we first look at the full URL, then at any Content-Type header, and finally at just the path portion of the URL. ASTERISK-27871 #close Change-Id: I16d0682f6d794be96539261b3e48f237909139cb	4 years ago
Sean Bright	785e4afc20	main/cdr.c: Correct Party A selection. ... This appears to just have been a copy/paste error from 6258bbe7. Fix suggested by Ross Beer in ASTERISK~29166. Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37	4 years ago
Sebastien Duthil	8a21d466ea	stun: Emit warning message when STUN request times out ... Without this message, it is not obvious that the reason is STUN timeout. ASTERISK-29507 #close Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74	4 years ago
Naveen Albert	244491f9b2	app_reload: New Reload application ... Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	4 years ago
Igor Goncharovsky	99d44f0c5a	res_ari: Fix audiosocket segfault ... Add check that data parameter specified when audiosocket used for externalMedia. ASTERISK-29514 #close Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617	4 years ago
Sean Bright	0ac9c83561	res_pjsip_config_wizard.c: Add port matching support. ... In f8b0c2c9 we added support for port numbers in 'match' statements but neglected to include that support in the PJSIP config wizard. The removed code would have also prevented IPv6 addresses from being successfully used in the config wizard as well. ASTERISK-29503 #close Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d	4 years ago
Naveen Albert	c01b4e0d4b	app_waitforcond: New application ... While several applications exist to wait for a certain event to occur, none allow waiting for any generic expression to become true. This application allows for waiting for a condition to become true, with configurable timeout and checking interval. ASTERISK-29444 Change-Id: I08adf2824b8bc63405778cf355963b5005612f41	4 years ago
Andre Barbosa	a47308ccb2	res_stasis_playback: Send PlaybackFinish event only once for errors ... When we try to play a list of sound files in the same Play command, we get only one PlaybackFinish event, after all sounds are played. But in the case where the Play fails (because channel is destroyed for example), Asterisk will send one PlaybackFinish event for each sound file still to be played. If the list is big, Asterisk is sending many events. This patch adds a failed state so we can understand that the play failed. On that case we don't send the event, if we still have a list of sounds to be played. When we reach the last sound, we send the PlaybackFinish with the failed state. ASTERISK-29464 #close Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322	4 years ago
George Joseph	bc973bd719	jitterbuffer: Correct signed/unsigned mismatch causing assert ... If the system time has stepped backwards because of a time adjustment between the time a frame is timestamped and the time we check the timestamps in abstract_jb:hook_event_cb(), we get a negative interval, but we don't check for that there. abstract_jb:hook_event_cb() then calls fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed) and the first thing that does is assert(interval >= 0). There are several issues with this... * abstract_jb:hook_event_cb() saves the interval in a variable named "now" which is confusing in itself. * "now" is defined as an unsigned int which converts the negative value returned from ast_tvdiff_ms() to a large positive value. * fixed_jb_get()'s parameter is defined as a signed int so the interval gets converted back to a negative value. * fixed_jb_get()'s assert is NOT an ast_assert but a direct define that points to the system assert() so it triggers even in production mode. So... * hook_event_cb()'s "now" was renamed to "relative_frame_start" and changed to an int64_t. * hook_event_cb() now checks for a negative value right after retrieving both the current and framedata timestamps and just returns the frame if the difference is negative. * fixed_jb_get()'s local define of ASSERT() was changed to call ast_assert() instead of the system assert(). ASTERISK-29480 Reported by: Dan Cropp Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9	4 years ago
Naveen Albert	1e5a2cfe30	app_dial: Expanded A option to add caller announcement ... Hitherto, the A option has made it possible to play audio upon answer to the called party only. This option is expanded to allow for playback of an audio file to the caller instead of or in addition to the audio played to the answerer. ASTERISK-29442 Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e	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
Bernd Zobl	c30f68a57b	res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress ... With the fix for ASTERISK_28754 channels are no longer put on hold if an outbound INVITE is answered with a "Session Progress" containing "inactive" audio. The previous change moved the evaluation of the media attributes to `negotiate_incoming_sdp_stream()` to have the `remotely_held` status available when building the SDP in `create_outgoing_sdp_stream()`. This however means that an answer to an outbound INVITE, which does not traverse `negotiate_incoming_sdp_stream()`, cannot set the `remotely_held` status anymore. This change moves the check so that both, `negotiate_incoming_sdp_stream()` and `apply_negotiated_sdp_stream()` can do the checks. ASTERISK-29479 Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75	4 years ago
George Joseph	b7027de195	res_pjsip_messaging: Overwrite user in existing contact URI ... When the MessageSend destination is in the form PJSIP/<number>@<endpoint> and the endpoint's contact URI already has a user component, that user component will now be replaced with <number> when creating the request URI. ASTERISK_29404 Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5	4 years ago
Bernd Zobl	f160725fc4	res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter ... Set preferred transport when querying the local address to use in filter_on_tx_messages(). This prevents the module to erroneously select the wrong transport if more than one transports of the same type (TCP or TLS) are configured. ASTERISK-29241 Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6	4 years ago
Naveen Albert	f812c57477	pbx_builtins: Corrects SayNumber warning ... Previously, SayNumber always emitted a warning if the caller hung up during execution. Usually this isn't correct, so check if the channel hung up and, if so, don't emit a warning. ASTERISK-29475 Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594	4 years ago
Jaco Kroon	56c2cc474b	func_lock: Add "dialplan locks show" cli command. ... For example: arthur*CLI> dialplan locks show func_lock locks: Name Requesters Owner uls-autoref 0 (unlocked) 1 total locks listed. Obviously other potentially useful stats could be added (eg, how many times there was contention, how many times it failed etc ... but that would require keeping the stats and I'm not convinced that's worth the effort. This was useful to troubleshoot some other issues so submitting it. Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	19a8383a1f	func_lock: Prevent module unloading in-use module. ... The scenario where a channel still has an associated datastore we cannot unload since there is a function pointer to the destroy and fixup functions in play. Thus increase the module ref count whenever we allocate a datastore, and decrease it during destroy. In order to tighten the race that still exists in spite of this (below) add some extra failure cases to prevent allocations in these cases. Race: If module ref is zero, an LOCK or TRYLOCK is invoked (near) simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if in such a case the datastore is created *prior* to unloading being set to true (first step in module unload) then it's possible that the module will unload with the destructor being called (and segfault) post the module being unloaded. The module will however wait for such locks to release prior to unloading. If post that we can recheck the module ref before returning the we can (in theory, I think) eliminate the last of the race. This race is mostly theoretical in nature. Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	e8875d5ca1	func_lock: Fix memory corruption during unload. ... AST_TRAVERSE accessess current as current = current->(field).next ... and since we free current (and ast_free poisons the memory) we either end up on a ast_mutex_lock to a non-existing lock that can never be obtained, or a segfault. Incidentally add logging in the "we have to wait for a lock to release" case, and remove an ineffective statement that sets memory that was just cleared by ast_calloc to zero. Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Jaco Kroon	caceba7988	func_lock: Fix requesters counter in error paths. ... In two places we bail out with failure after we've already incremented the requesters counter, if this occured then it would effectively result in unload to wait indefinitely, thus preventing clean shutdown. Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Naveen Albert	b742514553	app_originate: Allow setting Caller ID and variables ... Caller ID can now be set on the called channel and Variables can now be set on the destination using the Originate application, just as they can be currently using call files or the Manager Action. ASTERISK-29450 Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66	4 years ago
Sean Bright	c0fc8adbb6	menuselect: Fix description of several modules. ... The text description needs to be the last thing on the AST_MODULE_INFO line to be pulled in properly by menuselect. Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832	4 years ago
Naveen Albert	35437879e5	app_confbridge: New ConfKick() application ... Adds a new ConfKick() application, which may be used to kick a specific channel, all channels, or all non-admin channels from a specified conference bridge, similar to existing CLI and AMI commands. ASTERISK-29446 Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b	4 years ago
Naveen Albert	1b38e89734	res_pjsip_dtmf_info: Hook flash ... Adds hook flash recognition support for application/hook-flash. ASTERISK-29460 Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea	4 years ago
Naveen Albert	5f8cabc232	app_confbridge: New option to prevent answer supervision ... A new user option, answer_channel, adds the capability to prevent answering the channel if it hasn't already been answered yet. ASTERISK-29440 Change-Id: I26642729d0345f178c7b8045506605c8402de54b	4 years ago
Naveen Albert	c8bf8a54c2	sip_to_pjsip: Fix missing cases ... Adds the "auto" case which is valid with both chan_sip dtmfmode and chan_pjsip's dtmf_mode, adds subscribecontext to subscribe_context conversion, and accounts for cipher = ALL being invalid. ASTERISK-29459 Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2	4 years ago
George Joseph	c3654a9959	res_pjsip_messaging: Refactor outgoing URI processing ... * Implemented the new "to" parameter of the MessageSend() dialplan application. This allows a user to specify a complete SIP "To" header separate from the Request URI. * Completely refactored the get_outbound_endpoint() function to actually handle all the destination combinations that we advertized as supporting. * We now also accept a destination in the same format as Dial()... PJSIP/number@endpoint * Added lots of debugging. ASTERISK-29404 Reported by Brian J. Murrell Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce	4 years ago
Naveen Albert	eeffad1b62	func_math: Three new dialplan functions ... Introduces three new dialplan functions, MIN and MAX, which can be used to calculate the minimum or maximum of up to two numbers, and ABS, an absolute value function. ASTERISK-29431 Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d	4 years ago
Ben Ford	12e8600849	STIR/SHAKEN: Add Date header, dest->tn, and URL checking. ... STIR/SHAKEN requires a Date header alongside the Identity header, so that has been added. Still on the outgoing side, we were missing the dest->tn section of the JSON payload, so that has been added as well. Moving to the incoming side, URL checking has been added to the public cert URL to ensure that it starts with http. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c	4 years ago
Joshua C. Colp	44fde9f428	res_pjsip: On partial transport reload also move factories. ... For connection oriented transports PJSIP uses factories to produce transports. When doing a partial transport reload we need to also move the factory of the transport over so that anything referencing the transport (such as an endpoint) has the factory available. ASTERISK-29441 Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161	4 years ago
Naveen Albert	19b5097d87	func_volume: Add read capability to function. ... Up until now, the VOLUME function has been write only, so that TX/RX values can be set but not read afterwards. Now, previously set TX/RX values can be read later. ASTERISK-29439 Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f	4 years ago
Evgenios_Greek	2193cf1b26	stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing ... When unsubscribing from an endpoint technology a FRACK would occur due to incorrect reference counting. This fixes that issue, along with some other issues. Fixed a typo in get_subscription when calling ao2_find as it needed to pass the endpoint ID and not the entire object. Fixed scenario where a subscription would get returned when it shouldn't have been when searching based on endpoint technology. A doulbe unreference has also been resolved by only explicitly releasing the reference held by tech_subscriptions. ASTERISK-28237 #close Reported by: Lucas Tardioli Silveira Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729	4 years ago
Joseph Nadiv	98e4119642	res_pjsip.c: Support endpoints with domain info in username ... In multidomain environments, it is desirable to create PJSIP endpoints with the domain info in the endpoint name in pjsip_endpoint.conf. This resulted in an error with registrations, NOTIFY, and OPTIONS packet generation. This commit will detect if there is an @ in the endpoint identifier and generate the URI accordingly so NOTIFY and OPTIONS From headers will generate correctly. ASTERISK-28393 Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619	4 years ago
Joshua C. Colp	a985e5069c	res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates. ... RTCP ICE candidates use a base address derived from the RTP candidate. The port on the base address was not being updated to the RTCP port. This change sets the base port to the RTCP port and all is well. ASTERISK-29433 Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040	4 years ago
Joshua C. Colp	987f5eb0ad	asterisk: We've moved to Libera Chat! ... Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575	4 years ago
Jeremy Lainé	d162789c4d	res_rtp_asterisk: make it possible to remove SOFTWARE attribute ... By default Asterisk reports the PJSIP version in a SOFTWARE attribute of every STUN packet it sends. This may not be desired in a production environment, and RFC5389 recommends making the use of the SOFTWARE attribute a configurable option: https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2 This patch adds a `stun_software_attribute` yes/no option to make it possible to omit the SOFTWARE attribute from STUN packets. ASTERISK-29434 Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b	4 years ago
George Joseph	9cc1d6fc22	res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs ... RFC7616 and RFC8760 allow more than one WWW-Authenticate or Proxy-Authenticate header per realm, each with different digest algorithms (including new ones like SHA-256 and SHA-512-256). Thankfully however a UAS can NOT send back multiple Authenticate headers for the same realm with the same digest algorithm. The UAS is also supposed to send the headers in order of preference with the first one being the most preferred. We're supposed to send an Authorization header for the first one we encounter for a realm that we can support. The UAS can also send multiple realms, especially when it's a proxy that has forked the request in which case the proxy will aggregate all of the Authenticate headers and then send them all back to the UAC. It doesn't stop there though... Each realm can require a different username from the others. There's also nothing preventing each digest algorithm from having a unique password although I'm not sure if that adds any benefit. So now... For each Authenticate header we encounter, we have to determine if we support the digest algorithm and, if not, just skip the header. We then have to find an auth object that matches the realm AND the digest algorithm or find a wildcard object that matches the digest algorithm. If we find one, we add it to the results vector and read the next Authenticate header. If the next header is for the same realm AND we already added an auth object for that realm, we skip the header. Otherwise we repeat the process for the next header. In the end, we'll have accumulated a list of credentials we can pass to pjproject that it can use to add Authentication headers to a request. NOTE: Neither we nor pjproject can currently handle digest algorithms other than MD5. We don't even have a place for it in the ast_sip_auth object. For this reason, we just skip processing any Authenticate header that's not MD5. When we support the others, we'll move the check into the loop that searches the objects. Changes: * Added a new API ast_sip_retrieve_auths_vector() that takes in a vector of auth ids (usually supplied on a call to ast_sip_create_request_with_auth()) and populates another vector with the actual objects. * Refactored res_pjsip_outbound_authenticator_digest to handle multiple Authenticate headers and set the stage for handling additional digest algorithms. * Added a pjproject patch that allows them to ignore digest algorithms they don't support. This patch has already been merged upstream. * Updated documentation for auth objects in the XML and in pjsip.conf.sample. * Although res_pjsip_authenticator_digest isn't affected by this change, some debugging and a testsuite AMI event was added to facilitate testing. Discovered during OpenSIPit 2021. ASTERISK-29397 Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281	4 years ago
Joseph Nadiv	3cccdf6d98	res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml ... RFC 4235 Section 4.1.6 describes XML elements that should be sent to subscribed endpoints to identify the local and remote participants in the dialog. This patch adds this functionality to PJSIP by iterating through the ringing channels causing the NOTIFY, and inserts the channel info into the dialog so that information is properly passed to the endpoint in dialog-info+xml. ASTERISK-24601 Patch submitted: Joshua Elson Modified by: Joseph Nadiv and Sean Bright Tested by: Joseph Nadiv Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b	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
Naveen Albert	567ea5abf8	app_voicemail: Configurable voicemail beep ... Hitherto, VoiceMail() played a non-customizable beep tone to indicate the caller could leave a message. In some cases, the beep may not be desired, or a different tone may be desired. To increase flexibility, a new option allows customization of the tone. If the t option is specified, the default beep will be overridden. Supplying an argument will cause it to use the specified file for the tone, and omitting it will cause it to skip the beep altogether. If the option is not used, the default behavior persists. ASTERISK-29349 Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280	4 years ago