724 Commits (c47dbf1fbeaec678c7b2dbba51ebb13eaa286a8e)

Author	SHA1	Message	Date
Kevin Harwell	c47dbf1fbe	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Sean Bright	5f28ecc7af	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	6 years ago
George Joseph	4d6346dc26	doc: Fix CHANGES entries to have .txt suffix and update READMEs ... Although the wiki page for the new CHANGES and UPGRADE scheme states that the files must have the ".txt" suffix, the READMEs didn't. Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05 (cherry picked from commit 7416703f04)	6 years ago
George Joseph	54cd865946	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Asterisk Development Team	a0f44b1052	Update CHANGES and UPGRADE.txt for 17.2.0	6 years ago
Sean Bright	1e79bf44fc	http: Add ability to disable /httpstatus URI ... Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	6 years ago
Sean Bright	9ff062f994	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Sean Bright	a810025782	func_curl: Add 'followlocation' option to CURLOPT() ... We allow for 'maxredirs' to be set, but this value is ignored when followlocation is not enabled which, by default, it is not. ASTERISK-17491 #close Reported by: candrews Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a	6 years ago
Sean Bright	eac6eb663d	res_pjsip_endpoint_identifier_ip.c: Add port matching support ... Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	6 years ago
Friendly Automation	0b09aa087c	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 17	6 years ago
Friendly Automation	73131cefaf	Merge "app_dial.c: Simplify dialplan using Dial." into 17	6 years ago
Richard Mudgett	62a8750d2d	app_chanisavail.c: Simplify dialplan using ChanIsAvail. ... Dialplan has to be careful about passing an empty device list or empty positions in the list. As a result, dialplan has to check for these conditions before using ChanIsAvail. Simplify dialplan by making ChanIsAvail handle these conditions gracefully. * Made tolerate empty positions in the device list. * Simplified the code and eliminated some unnecessary indention. ASTERISK-28638 Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3	6 years ago
Richard Mudgett	b3d0b0c484	app_dial.c: Simplify dialplan using Dial. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Dial. Simplify dialplan by making Dial handle these conditions gracefully. * Made tolerate empty positions in the dialed device list. * Reduced some message log levels from notice to verbose. ASTERISK-28638 Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9	6 years ago
Richard Mudgett	f7c6a7df9f	app_page.c: Simplify dialplan using Page. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	6 years ago
Friendly Automation	b401891036	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 17	6 years ago
Joshua C. Colp	41fdcf7a65	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
Friendly Automation	a7a5bf0129	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
Kevin Harwell	68724425e4	res_fax: wrap v21 detected Asterisk initiated negotiation with config option ... A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
Asterisk Development Team	08a6e8c553	Update CHANGES and UPGRADE.txt for 17.1.0	6 years ago
Pascal Cadotte Michaud	39aff53cff	PJSIP_CONTACT: add missing argument documentation ... add missing argument "rtt" and "status" to the documentation The change to the dtd file allow an enumlist to contain one or many configOptionToEnum or enum. This is different from the previous patch I submitted when you could have a configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or (one or many enums) ASTERISK-28626 Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d	6 years ago
George Joseph	87d5fb1196	Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" into 17	6 years ago
Joshua Colp	6a89784b78	Revert "PJSIP_CONTACT: add missing argument documentation" ... This reverts commit 450173a0ae. Reason for revert: Regression in XML validation. validity error : Content model of enumlist is not determinist: (configOptionToEnum | (configOptionToEnum , enum+) | enum+) As we are preparing to do releases and this is not critical I am reverting this for now until resolved. Change-Id: I47a6714bb555b35a371861d434daf5b97c0cdcbd	6 years ago
Friendly Automation	e17bde758b	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
George Joseph	6b1ba58967	manager.c: Prevent the Originate action from running the Originate app ... If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	6 years ago
Pascal Cadotte Michaud	450173a0ae	PJSIP_CONTACT: add missing argument documentation ... add missing argument "rtt" and "status" to the documentation ASTERISK-28626 Change-Id: I8419e4c8203e411b87d93dc395acdbcf7526dedf	6 years ago
Martin Tomec	d579ec9cdf	func_curl.c: Support custom http headers ... When user wants to send json data, the default Content-Type header is incorect (application/x-www-form-urlencoded). This patch allows to set any custom headers so the Content-Type header can be overriden. User can set multiple headers by multiple calls of curlopt(). This approach is not consistent with other parameters, but is more readable in dialplan than one call with multiple headers. ASTERISK-28613 Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9	6 years ago
Sean Bright	b903994987	Revert "app_voicemail: Cleanup stale lock files on module load" ... This reverts commit fd2e8d0da7. Reason for revert: Problematic for users who store their voicemail on network storage devices, or share voicemail storage between multiple Asterisk instances. ASTERISK-28567 #close Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41	6 years ago
Torrey Searle	55b760d762	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
George Joseph	a398196fd0	Merge "res_musiconhold: Add new 'playlist' mode" into 17	6 years ago
Ben Ford	8269fcbf03	taskprocessor.c: Added "like" support to 'core show taskprocessors' ... Added "like" support for 'core show taskprocessors'. Now you can specify a specific set of taskprocessors (or just one) by adding the keyword "like" to the above command, followed by your search criteria. Change-Id: I021e740201e9ba487204b5451e46feb0e3222464	6 years ago
Sean Bright	7550a82fe0	res_musiconhold: Add new 'playlist' mode ... Allow the list of files to be played to be provided explicitly in the music class's configuration. The primary driver for this change is to allow URLs to be used for MoH. Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa	6 years ago
Ben Ford	f181dfc065	taskprocessor.c: Add CLI commands to reset taskprocessor stats. ... Added two new CLI commands to reset stats for taskprocessors. You can reset stats for a single, specific taskprocessor ('core reset taskprocessor <taskprocessor>'), or you can reset all taskprocessors ('core reset taskprocessors'). These commands will reset the counter for the number of tasks processed as well as the max queue size. Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d	6 years ago
Joshua Colp	6e75e2565d	Merge "func_jitterbuffer: Add audio/video sync support." into 17	6 years ago
Joshua Colp	926053d7bd	func_jitterbuffer: Add audio/video sync support. ... This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492	6 years ago
Florian Floimair	d7a3e4f5cf	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
sungtae kim	9e26136ee6	res_musiconhold: Added unregister realtime moh class ... This fix allows a realtime moh class to be unregistered from the command line. This is useful when the contents of a directory referenced by a realtime moh class have changed. The realtime moh class is then reloaded on the next request and uses the new directory contents. ASTERISK-17808 Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce	6 years ago
George Joseph	44068cc6bf	ARI: External Media ... The Channel resource has a new sub-resource "externalMedia". This allows an application to create a channel for the sole purpose of exchanging media with an external server. Once created, this channel could be placed into a bridge with existing channels to allow the external server to inject audio into the bridge or receive audio from the bridge. See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI for more information. Change-Id: I9618899198880b4c650354581b50c0401b58bc46	6 years ago
George Joseph	581ffdf6a9	chan_rtp: Accept hostname as well as ip address as destination ... The UnicastRTP channel driver provided by chan_rtp now accepts "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination. The first AAAA (preferred) or A record resolved will be used as the destination. The lookup is synchronous so beware of possible dialplan delays if you specify a hostname. Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677	7 years ago
Sean Bright	e123d1ccc5	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	7 years ago
Asterisk Development Team	f1e0c9d1b8	Update CHANGES and UPGRADE.txt for 17.0.0	7 years ago
George Joseph	4f2ad5a55d	doc: Add "master-only" flag back to the CHANGES and UPGRADE files ... In order to run the documentation scripts the flags needs to be added back to the staging files. Change-Id: Ia10a153c50c970cfa1e85815208dfaddb3f2ccd4	7 years ago
George Joseph	80d8dce6af	Prepare Asterisk 17 Branch ... Change-Id: Idb79a69646d2511e7bf1573b9b0322cc22ea54e8	7 years ago
George Joseph	799c4cf494	Merge "chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS"	7 years ago
George Joseph	c781806e26	Build: Separate header install/uninstall ... Asterisk headers are no longer installed and uninstalled automatically when performing a "make install" or a "make uninstall". To install/uninstall the headers, use "make install-headers" and "make uninstall-headers". The headers also continue to be uninstalled when performing a "make uninstall-all". Also corrects an issue where /usr/include/asterisk.h was never being removed at all. Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643	7 years ago
Chris-Savinovich	6b1f6ea2c4	app_voicemail.c: Build all three variants for app_voicemail at the same time ... Changes made to apps/Makefile to optionally build all three app_voicemail variations at the same time: 1) file (default), 2) odbc, and 3) imap. This functionality was requested by users. modules.conf.sample warns the user to make sure only one voicemail is loaded at a time. Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7	7 years ago
Dan Cropp	e52fbae00f	chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS ... Previously, when a Transfer (REFER) was performed, chan_pjsip would set the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not reflect a successful/unsuccessful transfer the way chan_sip did. Added a callback module to process the refer subscription information. Now depends on res_pjsip_pubsub so call transfer progress can be monitored and reported ASTERISK-26968 #close Reported-by: Dan Cropp Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc	7 years ago
Joshua Colp	a8e5cf557d	res_rtp_asterisk: Add support for DTLS packet fragmentation. ... This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	7 years ago
George Joseph	93ccff25c6	Merge "app_attended_transfer: new application AttendedTransfer"	7 years ago
Friendly Automation	9bb9700e07	Merge "app_blind_transfer: new application BlindTransfer"	7 years ago
Alexei Gradinari	3eaeb3e6c4	app_attended_transfer: new application AttendedTransfer ... AttendedTransfer queues up attended transfer to the given extension. This application can be useful with Custom Dynamic Features. For example to make attended transfer to a predefined number. features.conf ;;; [applicationmap] my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_atxfer TRANSFER_CONTEXT=my_transfer [my_atxfer] exten => s,1,AttendedTransfer(1234567890) same => n,Return() [my_transfer] include => default ;;; This application also can be used to completly redefine Attended transfer feature using dialplan. For example: features.conf ;;; [featuremap] atxfer => *7 [applicationmap] custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_atxfer TRANSFER_CONTEXT=my_transfer [custom_atxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,AttendedTransfer(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	7 years ago