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Author	SHA1	Message	Date
Richard Mudgett	fa3370fdee	Some CDR code optimization. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7e3c893bb8	Whitespace and curly braces. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eb36b71183	Improve detection of answer on SIP blind transfer. ... A problem encountered during testing was that res_pjsip_refer would not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook that was supposed to send the NOTIFY would never be told that an answer had occurred. This happened for two reasons: 1) The transferee channel on which the framehook was on was already up. 2) Answers are rarely if ever written to channels. Rather, the ast_answer() or ast_raw_answer() function is used to answer channels. Thanks to a suggestion by Matt Jordan, the best way to detect that the call had been answered was to find out when the transferee channel joined a bridge. With stasis this is an easy task. So now, in addition to the framehook logic, there is a stasis subscription used to determine when the transferee has entered a bridge. Once it has entered, an appropriate NOTIFY is sent. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	402d4578ba	Add database schema management using Alembic ... This patch replaces contrib/realtime/ with a new setup for managing the database schema required for database integration with Asterisk. In addition to initializing a database with the proper schema, alembic can do a database migration to assist with upgrading Asterisk in the future. Hopefully this helps make setting up and operating Asterisk with a database easier. With this the schema only needs to be maintained in one place instead of once per database. The schemas I have added here have a bit of improvement over the examples that were there before (some added consistency and added some missing indexes). Managing the schema in one place here also applies to all databases supported by SQLAlchemy. See contrib/ast-db-manage/README.md for more details. Review: https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant (license 6300) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0f61ca1b36	Update CHANGES file for Asterisk 12 ... This updates the Asterisk 12 CHANGES file with the things that were missed during the development cycle. Review: https://reviewboard.asterisk.org/r/2795/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ae6e4d27e2	pbx.c: Make ast_str_substitute_variables_full() not mask variables. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e19173ca53	Match use of ast_free() with ast_calloc() and add some curly braces. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	81f63aa2dc	Fix dialog matching in the SIP distributor. ... Dialog matching is performed in the distributor for the sole purpose of retrieving an associated serializer so the request may be serialized. This patch fixes two problems. First, incoming CANCEL requests that had no to-tag (which really should be *all* CANCEL requests) would not match with a dialog. An earlier bug fix to deal with early CANCEL requests would result in the CANCEL being replied to with a 481. The fix for this is to find the matching INVITE transaction and get the dialog from that transaction. Second, no SIP responses were matching dialogs. This is because we were inverting the tags that we were passing into PJSIP's dialog finding function. This logic has been corrected by setting local and remote tag variables based on whether the incoming message is a request or response. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	d7e16fadf8	ARI: WebSocket event cleanup ... Stasis events (which get distributed over the ARI WebSocket) are created by subscribing to the channel_all_cached and bridge_all_cached topics, filtering out events for channels/bridges currently subscribed to. There are two issues with that. First was a race condition, where messages in-flight to the master subscribe-to-all-things topic would get sent out, even though the events happened before the channel was put into Stasis. Secondly, as the number of channels and bridges grow in the system, the work spent filtering messages becomes excessive. Since r395954, individual channels and bridges have caching topics, and can be subscribed to individually. This patch takes advantage, so that channels and bridges are subscribed to on demand, instead of filtering the global topics. The one case where filtering is still required is handling BridgeMerge messages, which are published directly to the bridge_all topic. Other than the change to how subscriptions work, this patch mostly just moves code around. Most of the work generating JSON objects from messages was moved to .to_json handlers on the message types. The callback functions handling app subscriptions were moved from res_stasis (b/c they were global to the model) to stasis/app.c (b/c they are local to the app now). (closes issue ASTERISK-21969) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f429557bf8	Made MALLOC_DEBUG less CPU intensive by default. ... Storing a backtrace for each allocation in anticipation of a memory management problem is very CPU intensive. * Added the CLI "memory backtrace {on|off}" command to request that the backtrace be gathered only on request. The backtrace is off by default. (issue ASTERISK-22221) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	54d8f36572	AST-2013-005: Fix crash caused by invalid SDP ... If the SIP channel driver processes an invalid SDP that defines media descriptions before connection information, it may attempt to reference the socket address information even though that information has not yet been set. This will cause a crash. This patch adds checks when handling the various media descriptions that ensures the media descriptions are handled only if we have connection information suitable for that media. Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing the solution to this problem. (closes issue ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 397758 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1072858d0f	Fix uninitialized value in struct ast_control_pvt_cause_code usage. ... ........ Merged revisions 397744 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5e34c49a66	AST-2013-004: Fix crash when handling ACK on dialog that has no channel ... A remote exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumed that the channel would always be present. This patch adds a check such that the SDP will only be parsed and applied if Asterisk has a channel present that is associated with the dialog. Note that the patch being applied was modified only slightly from the patch provided by Walter Doekes of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 397712 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ff68143bd0	Better handle clearing the OUTGOING flag when a channel leaves a bridge ... When a channel with the OUTGOING flag leaves a bridge, and it will survive being pulled from the bridge (either because it will execute dialplan, go into another bridge, or live in a friendly autoloop), we have to clear the OUTGOING flag. This is the signal to the CDR engine that this channel is no longer a second class citizen, i.e., it is not "dialed". The soft hangup flags are only half the picture. If a channel is being moved from one bridge to another, the soft hangup flags aren't set; however, the state of the bridge_channel will not be hung up. Since the channel does not have one of the two hang up states, that implies that the channel is still technically alive. This patch modifies the check so that it checks both the soft hangup flags as well as the bridge_channel state. If either suggests that the channel is going to persist, we clear the OUTGOING flag. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	594ef4e5ab	Fixed bucket.c for systems where tv_usec is not an unsigned long. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7a60a318ce	bridging: Fix a livelock with local channel optimization. ... Use a better means of waking up the bridge channel thread. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2d57781191	chan_dahdi: Add some missing build cleanup. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	029057a261	Fix bucket unit tests ... After the review for buckets was completed (r2715), the handling of names in the bucket core was deferred to the wizards. As such, the bucket unit tests cannot expect that passing a URI with a scheme specified but no actual resource name will automatically fail. The tests have been updated to not make this check. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1888186add	Fix the config_options_test ... The config options test requires the entire configuration item to be transparent from the documentation system. So we let it do that too. As an aside, please do not use this power for evil. Documentation is your friend, and you really should document your configurations. Hiding your module's configuration information from the system attempting to enforce some sanity in the universe is something only a Bond villain would contemplate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	206f07a721	Add rtpengine configuration parameter ... The rtpengine configuration parameter was documented in the XML documentation, but it was not actually registered with the sorcery object. This adds the parameter with a default of "asterisk", such that res_rtp_asterisk is chosen as the default RTP implementation. (closes issue ASTERISK-22380) Reported by: Rusty Newton Tested by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	006c0f4b5e	__________ ... | \ |_______ | | | ______| | / | _ _ _ _ _ | _______| / \ ___| |_ ___ _ __(_)___| | __ / || | / _ \ / __| __/ _ \ '__| / __| |/ / | || |_______ / ___ \__ \| | __/ | | \__ \ < | || | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| \__________| git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	9c713d12da	Fix building of trunk. ... Note: This is why I commit on the weekend. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f4bf1823e9	Fix channel reference leak in Originated channels ... When originating channels, ast_pbx_outgoing_* caused the dialed channel reference to be bumped twice. Ostensibly, this routine is bumping the channel lifetime such that the channel doesn't get nuked in between locks/unlocks; however, since the routine should return the dialed channel with its reference bumped, it only needs to do this one time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	00d8ef8c79	Blocked revisions 397604 ... ........ Make libuuid an optional dependency for res_rtp_asterisk instead of a requirement. Review: https://reviewboard.asterisk.org/r/2777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e7c6f33517	Add some clarifying documentation to the rewrite_contact endpoint option. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fd7ac4fecd	Blank line tweaks. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	dd33217762	Add the bucket API. ... Bucket is a URI based API for the creation, retrieval, updating, and deletion of "buckets" and files contained within them. Review: https://reviewboard.asterisk.org/r/2715/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cd7436a6c0	Fix a bug where the argc value was passed as no_doc when registering custom sorcery types. ... This also adds a _nodoc equivalent. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	cae6bc54c8	Add test events necessary for bridge tests to pass in the test suite. ... (closes issue AST-1200) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2790/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3d9c5e61af	Fix error in using ast_channel_snapshot_type before initialization ... Starting Asterisk would kick back an ERROR message stating that the Stasis message type ast_channel_snapshot_type was used prior to initialization. This occurred due to the caching topic being created prior to the message type that it depended on. This patch re-orders the start up such that the message type is initialized prior to the caching topic. It also checks the return value of the initialization of the agent login/logoff types. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c13d0b7bdc	bridge_native_rtp: Fix hold chain bugs caused by native RTP bridge framehook ... Issuing hold/unhold would lead to odd behavior. Between two chan_sip devices, a hold could cause an endless chain of updates while with pjsip a similar chain would begin but then end somewhat randomly. This patch fixes that by no longer tweaking the RTP glue on both sides of the call for every HOLD/UNHOLD/UPDATE_RTP_PEER frame. (issue ASTERISK-22217) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2794/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6ebfac8e70	Handle DTMF and hold wrapup when a channel leaves the bridging system. ... DTMF start/end and hold/unhold events have state because a DTMF begin event and hold event must be ended by something. The following cases need to be handled when a channel is moved around in the system. * When a channel leaves a bridge it may owe a DTMF end event to the bridge. * When a channel leaves a bridge it may owe an UNHOLD event to the bridge. (This case is explicitly ignored because things like transfers need explicit control over this.) * When a channel leaves the bridging system it may need to simulate a DTMF end event to the channel. * When a channel leaves the bridging system it may need to simulate an UNHOLD event to the channel. The patch also fixes the following: * Fixes playing a file and restarting MOH using the latest MOH class used. (closes issue ASTERISK-22043) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2791/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	31ab486627	Fix sorcery unit tests ... When strict XML documentation checking was re-enabled, the test objects used in sorcery would fail to register as the types were not marked internal and the nodoc option wasn't used for the options. This fixes that problem, such that, as one would hope, they once again pass. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	46b9e5450f	Fix memory corruption when trying to get "core show locks". ... Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch in memory pools but had a math error determining the buffer size and didn't address other similar memory pool mismatches. * Effectively reverted the previous patch to go in the same direction as trunk for the returned memory pool of ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is enabled. * Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of compile issues with the utils directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	32a0567c46	Prevent seg fault in off nominal path when registered option fails to validate ... If an option is registered to a type and it is the last known type in the list of registered types, and the option fails to register, an overrun of the types array can occur due to the index variable having been already incremented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	882ba86434	PSJIP - sip.conf to res_sip.conf script ... Most, if not all, of the backing features of a conf file should now be implemented (e.g. multi-line comments, includes, templates, etc...). A few of the options still need to be mapped. Those are currently listed in the 'sip_to_res_sip.py' file. Things to do: (1) There is more work to do here, at least for the sip.conf items that aren't currently parsed. An issue will be created for that. (2) All of the scripts should probably be passed through pylint and have as many PEP8 issues fixed as possible. (3) A public review is probably warranted at that point of the entire script. Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	7a581537e8	ARI: Correct error codes for bridge operations ... This patch adds error checking to ARI bridge operations, when adding/removing channels to/from bridges. In general, the error codes fall out as follows: * Bridge not found - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel not found - 400 Bad Request * Channel not in Stasis - 422 Unprocessable Entity * Channel not in this bridge (on remove) - 422 Unprocessable Entity (closes issue ASTERISK-22036) Review: https://reviewboard.asterisk.org/r/2769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9f4849724f	Update CHANGES file to reflect pass through support for Opus/VP8 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4d348e853c	Add pass through support for Opus and VP8; Opus format attribute negotiation ... This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e31bd332b8	Update config framework/sorcery with types/options without documentation ... There are times when a configuration option should not have documentation. 1. Some options are registered with a particular object merely as a warning to users. These options aren't even really 'deprecated' - which has its own separate API call - they are actually provided by a different configuration file. The options are merely registered so that the user gets a warning that a different configuration file provides the item. 2. Some object types - most notably some used by modules that use sorcery - are completely internal and should never be shown to the user. 3. Sorcery itself has several 'hidden' fields that should never be shown to a user. This patch updates the configuration framework and sorcery with additional API calls that allow a module to register types as internal and options as not requiring documentation. This bypasses the XML documentation checking. This patch also re-enables the strict XML documentation checking in trunk, as well as updates some documentation that was missing. Review: https://reviewboard.asterisk.org/r/2785/ (closes issue ASTERISK-22359) Reported by: Matt Jordan (closes issue ASTERISK-22112) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	b2a13e83dc	Fix crash when answering after a transport error occurs. ... If a response to an initial incoming INVITE results in a transport error the INVITE transaction is removed from the INVITE session. Any attempts to answer the INVITE session after this results in a crash as it requires the INVITE transaction to exist. This change explicitly locks the dialog and checks to ensure that the INVITE transaction exists before answering. (closes issue AST-1203) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	d12c79f78f	Update CEL sample config ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	21e22310c7	ARI: Music on Hold/Background Music for bridges ... Adds ARI functions to be able to turn on/off music on hold in a bridge. It actually functions more as a background music without further actions on the bridge since if the rest of the channels in the bridge aren't explicitly muted, they will still be able to communicate. (closes issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c25c093c67	Minor tweaks with ast_moh_start() callers. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	7b032c1adb	Add SayAlphaCase and similar functionality for AGI ... This adds a new dialplan application, SayAlphaCase, that performs much the same function as SayAlpha except that it takes additional options which allow the user to specify whether the case of each letter should be announced for uppercase, lowercase, or all letters. Similar functionality has been added to the SAY ALPHA AGI command via an optional parameter. Original Patch by: Kevin Scott Adams Reported by: Kevin Scott Adams Review: https://reviewboard.asterisk.org/r/2725/ (closes issue ASTERISK-20782) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	aefebebd37	res_sip_dtmf_info: Support sending of 'raw' DTMF ... Added the ability to handle 'raw' DTMF within the body of an INFO message. Also made it so values 10-16 are mapped to valid DTMF values. (closes issue ASTERISK-22144) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	6fa4e8e3ab	Add missing configOption close tags ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c02c1b6f53	Update MOH start/stop routine doxygen. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	9520df86fd	Fix missing xml doc configOption 'type' for for both 'system' and 'global' configObjects ... (issue ASTERISK-22344) (closes issue ASTERISK-22344) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	477dea4661	Bridge API: Set a cause code on a channel when it is ejected from a bridge. ... The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago