737 Commits (0ce4a773673c3639bb500801fae8d24b220e67b4)

Author	SHA1	Message	Date
Asterisk Development Team	bbd0835482	Update CHANGES and UPGRADE.txt for 16.10.0	6 years ago
Joshua C. Colp	2128eb1f47	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Joshua C. Colp	62183bc777	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	6 years ago
Jean Aunis	ef4255f6ed	func_volume: Accept decimal number as argument ... Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	6 years ago
Joshua C. Colp	99869810a1	CHANGES: Change md file extension to txt. ... Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7 (cherry picked from commit c9cd681261)	6 years ago
Joshua C. Colp	1c5129bca4	res_pjsip_session: Apply intention behind requested formats. ... When an outgoing channel is created a list of formats may optionally be provided which is used as a request that the formats be used if possible. If an endpoint is not configured for any of the formats we ignore this request and use what is configured. This has the side effect of also including other stream types (such as video) that were not present in the requested formats. This change makes it so that the intention of the request is preserved - that is if only an audio format is requested then even if there is no joint audio format between the request and the configuration we will still only place an audio stream in the outgoing call. ASTERISK-28787 Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f	6 years ago
Jaco Kroon	351b2be00a	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. ... A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Rodrigo Ramírez Norambuena	7dcabaef08	res_rtp_asterisk: Add 'rtp show settings' cli command ... This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	6 years ago
Asterisk Development Team	1522c4467c	Update CHANGES and UPGRADE.txt for 16.9.0	6 years ago
Kevin Harwell	fc1d93cf97	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	cd8b27dcc2	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	1544f74932	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	d6574cb7c7	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	eb1ec0498d	Update CHANGES and UPGRADE.txt for 16.8.0	6 years ago
Sean Bright	a2a4e1026c	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	f5a1e8b04d	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Sean Bright	f69da94fab	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	f8b0c2c933	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	7c227aa130	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 16	6 years ago
Friendly Automation	077bb05528	Merge "app_dial.c: Simplify dialplan using Dial." into 16	6 years ago
Richard Mudgett	a7692ce2f4	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	144b774b85	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	2780be334d	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	a992180091	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 16	6 years ago
Friendly Automation	a3edac10a6	Merge "confbridge: Add support for specifying maximum sample rate." into 16	6 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Friendly Automation	298dd7832b	Merge "PJSIP_CONTACT: add missing argument documentation" into 16	6 years ago
Kevin Harwell	d17bbcb9f1	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	9240fcd8bb	Update CHANGES and UPGRADE.txt for 16.7.0	6 years ago
Pascal Cadotte Michaud	2d2b28bfa4	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	4631e77078	Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" into 16	6 years ago
Joshua Colp	9c29c3fb3e	Revert "PJSIP_CONTACT: add missing argument documentation" ... This reverts commit 174e6426aa. 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: I2c9656fb40b2d2f56f54caa35c8be02cc92babd0	6 years ago
Friendly Automation	7620d1256c	Merge "PJSIP_CONTACT: add missing argument documentation" into 16	6 years ago
Asterisk Development Team	9eb86a8110	Update CHANGES and UPGRADE.txt for 16.6.2	6 years ago
George Joseph	7574be5110	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	174e6426aa	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	37dcdd485a	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	68855f6a7b	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	9a933c3adc	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	7 years ago
George Joseph	072c63f575	Merge "taskprocessor.c: Added "like" support to 'core show taskprocessors'" into 16	7 years ago
Friendly Automation	1903dcb43d	Merge "res_musiconhold: Add new 'playlist' mode" into 16	7 years ago
root	d146ec7e83	Revert "Update CHANGES and UPGRADE.txt for 16.6.0-rc2" ... This reverts commit a0844a9b9b. Correcting/reverting accidental upgrade on release process. Change-Id: I08746eda578effa6e64054690fad48a17d2f42e7	7 years ago
Ben Ford	827dd754b2	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	7 years ago
Asterisk Development Team	a0844a9b9b	Update CHANGES and UPGRADE.txt for 16.6.0-rc2	7 years ago
Ben Ford	5ea667e03a	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	7 years ago
Friendly Automation	a687c7919d	Merge "func_jitterbuffer: Add audio/video sync support." into 16	7 years ago
Joshua Colp	120f2fb5f9	Merge "core: Add H.265/HEVC passthrough support" into 16	7 years ago
Joshua Colp	6647be69ac	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	7 years ago
Sean Bright	9f304170f6	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	7 years ago
Asterisk Development Team	688908fe7a	Update CHANGES and UPGRADE.txt for 16.6.0	7 years ago