4656 Commits (4be231e82f840fb1cb2329698ba2184d8b726f2c)

Author	SHA1	Message	Date
Rodrigo Ramírez Norambuena	f2a84b500d	app_queue: Show reason of pause on CLI ... Add value of pause reason when is paused on CLI command "queue show" ASTERISK-25581 #close Report by: Rodrigo Ramírez Norambuena Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864	10 years ago
Joshua Colp	b52b494004	Merge "StatsD: Add sample rate compatibility"	10 years ago
Matt Jordan	65102d19bc	Merge "app_bridgeaddchan: ability to barge into existing call"	10 years ago
Alec Davis	8c14b91651	app_bridgeaddchan: ability to barge into existing call ... To be able to barge into a call by dialling a prefix+extension that maps to the extensions device. Senario is that DECT headset users may be away from their desks and need to transfer the call, the goal is that from any phone they dial a prefix then their extension and are added to the bridge that they are in, from there they can drop the headset call, as it's also on the handset, and transfer the caller. The dialplan would look like, where prefix=73, extension = 8512; exten => _738512,1,BridgeAdd(SIP/cisco0001) ASTERISK-25551 #close Reported By: Alec Davis Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540	10 years ago
tcambron	05addf3d8f	StatsD: Add sample rate compatibility ... Implemented support for the StatsD sample rate parameter, which is a parameter for determining when to send computed statistics to a client. Valid sample rate values are: Less than or equal to 0.0 will never be sent. Between 0.0 and 1.0 will randomly be sent. Greater than or equal to 1.0 will always be sent. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52	10 years ago
Alec Davis	4013f9d577	app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! ... commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a	10 years ago
Mark Michelson	ed13732188	Confbridge: Add a user timeout option ... This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Joshua Colp	0be147f713	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased."	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Walter Doekes	39daf9f066	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
Joshua Colp	29810aa620	Merge "StatsD: Send stuff to the StatsD server and test"	10 years ago
tcambron	c5093b21ad	StatsD: Send stuff to the StatsD server and test ... Added code to allow the StatsD dialplan application to send data to the server specified in statsd.conf. ASTERISK-25419 Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922	10 years ago
Matt Jordan	2ddd9ff2d9	Merge "app_queue: Added reason pause of member"	10 years ago
tcambron	fdfd0fb488	StatsD: Add user input validation to the application ... Added code to accept user input and validate it before allowing it to be sent to the StatsD server. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3	10 years ago
Rodrigo Ramírez Norambuena	92fa8d1e0e	app_queue: Added reason pause of member ... In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e	10 years ago
Matt Jordan	da54d3ddd1	Merge "StatsD: Write skeleton Asterisk application"	10 years ago
tcambron	2fe9f09705	StatsD: Write skeleton Asterisk application ... Wrote the skeleton framework for the Asterisk StatsD dialplan application. This includes a load function, unload function, a callback for execution, and XML documentation. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363	10 years ago
Matt Jordan	9db74be3c0	Merge "app_queue.c: Force COLP update if outgoing channel name changed."	10 years ago
Matt Jordan	8bb8f99252	Merge "app_queue.c: Factor out a connected line update routine."	10 years ago
Matt Jordan	e0d8b6a65d	Merge "app_dial.c: Make 'A' option pass COLP updates."	10 years ago
Matt Jordan	360d076dfc	Merge "app_dial.c: Force COLP update if outgoing channel name changed."	10 years ago
Joshua Colp	afabf9da7f	Merge "app_dial.c: Factor out a connected line update routine."	10 years ago
Richard Mudgett	7c7a7ddd27	app_queue.c: Force COLP update if outgoing channel name changed. ... * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 #close Reported by: John Hardin Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172	10 years ago
Richard Mudgett	145608bd81	app_queue.c: Factor out a connected line update routine. ... Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3	10 years ago
Richard Mudgett	1d394774b2	app_dial.c: Make 'A' option pass COLP updates. ... While the 'A' option is playing the announcement file allow the caller and peer to exchange COLP update frames. ASTERISK-25423 Reported by: John Hardin Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9	10 years ago
Richard Mudgett	680b76eb25	app_dial.c: Force COLP update if outgoing channel name changed. ... * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 Reported by: John Hardin Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c	10 years ago
Richard Mudgett	fdf0bcb04a	app_dial.c: Factor out a connected line update routine. ... Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091	10 years ago
Richard Mudgett	c285879845	app_dial.c: Remove some no-op code. ... Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54	10 years ago
Richard Mudgett	06f4f80a63	app_page.c: Fix crash when forwarding with a predial handler. ... Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981	10 years ago
Kevin Harwell	c74101509d	app_record: RECORDED_FILE variable not being populated ... The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653	10 years ago
Matt Jordan	9a4498a112	Merge "app_queue: AgentComplete event has wrong reason"	10 years ago
Kevin Harwell	729a4325da	app_queue: AgentComplete event has wrong reason ... When a queued caller transfers an agent to another extension sometimes the raised AgentComplete event has a reason of "caller" and sometimes "transfer". Since a transfer has taken place this should always be transfer. This occurs because sometimes the stasis hangup event arrives before the transfer event thus writing a different reason out. With this patch, when a hangup event is received during a transfer it will check to see if the channel that is hanging up is part of a transfer. If so it will return and let the subsequently received transfer event handler take care of the cleanup. ASTERISK-25399 #close Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d	10 years ago
Kevin Harwell	63ede41227	app_queue: Crash when transferring ... During some transfer scenarios involving queues Asterisk would sometimes crash when trying to obtain a channel snapshot (could happen on caller or member channels). This occurred because the underlying channel had already disappeared when trying to obtain the latest snapshot. This patch adds a reference to both the member and caller channels that extends to the lifetime of the queue'd call, thus making sure the channels will always exist when retrieving the latest snapshots. ASTERISK-25185 #close Reported by: Etienne Lessard Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128	10 years ago
Mark Michelson	26fca72837	Merge "app_queue.c: Extract some functions for simpler code."	10 years ago
Richard Mudgett	9fb4a96e15	app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. ... Setting the 'paused' and 'ringinuse' options on a queue member using the dialplan function QUEUE_MEMBER did not behave the same way as the equivalent dialplan applications or AMI actions. * Made queue_function_mem_write() call the set_member_paused() and set_member_value() for the 'paused' and 'ringinuse' options respectively. A beneficial side effect is that the queue name is now optional and sets the value in all queues the interface is a member. * Update QUEUE_MEMBER XML documentation. * Fix error checking in QUEUE_MEMBER() write. ASTERISK-25215 #close Reported by: Lorne Gaetz Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb	10 years ago
Richard Mudgett	87b22969a4	app_queue.c: Extract some functions for simpler code. ... * Extract set_queue_member_pause() from set_member_paused() for simpler and more consistent code. * Extract set_queue_member_ringinuse() from set_member_ringinuse_help_members() for simpler code. Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306	10 years ago
Richard Mudgett	5cf98e2459	app_queue.c: Fix error checking in QUEUE_MEMBER() read. ... Change-Id: I7294e13d27875851c2f4ef6818adba507509d224	10 years ago
Matt Jordan	eff6a88a88	apps/app_dictate: Fix typo in attribution ... Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago
Joshua Colp	3b2b004d69	app_dial: Hold reference to calling channel formats when dialing outbound. ... Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db	10 years ago
Richard Mudgett	a657ab12f9	app_directory: Fix crash when using the alias option 'a'. ... The voicemail.conf mailbox key/value pair is defined as: <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just <mailbox>=<password>,<name> when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374	11 years ago
Corey Farrell	80621ce3c5	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	11 years ago
George Joseph	31f0d78d7b	app_playback: Suppress warnings on playback if channel hung up ... If a channel hangs up while an audio file is playing, there's no need to clutter up the logs with a warning so suppress it if ast_check_hangup returns true. Also, change warning to debug/2 in file.c if writing a frame fails. Same reasoning. Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Joshua Colp	35ff01823b	Merge "AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro."	11 years ago
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. ... Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	11 years ago
Jonathan Rose	0d97d7cb94	app_voicemail: fix moving when old messages full ... When completing voicemail playback of a message in the 'INBOX', the message gets moved to the 'Old' messages folder. Without this patch, if the 'Old' folder is already at its set limit, then the 'INBOX' message will simply be deleted. With this patch, the flag to delete the message will be removed if the save_to_folder function indicates that the message could not be moved due to a full folder. ASTERISK-25082 #close Reported by: Jonathan Rose Review: https://gerrit.asterisk.org/#/c/448/ Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f	11 years ago
Ivan Poddubny	90bfc02e84	app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter ... This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3 parameters: position, original position and waiting time. ASTERISK-25038 #close Reported by: Etienne Lessard Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Kevin Harwell	9f65ea482e	app_confbridge: Default the template option to a compatible default profile. ... Confbridge dynamic profiles did not have a default profile unless you explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a template was not set prior to the bridge being created then some options were left with no default values set. This patch makes it so the default templates are set to the default bridge and user profiles. ASTERISK-24749 #close Reported by: philippebolduc Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a	11 years ago
Mark Michelson	aae45acbda	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	11 years ago
Corey Farrell	62508d6891	Build System: Create Makefile macro MOD_ADD_SOURCE. ... This new macro allows a single line to add all additional sources to a module. This helps prevent modules from missing steps, and makes future changes easier since they can be made in a single place. ASTERISK-24960 #close Reported by: Corey Farrell Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b	11 years ago