asterisk

Commit Graph

Author	SHA1	Message	Date
Jean Aunis	de66713fd5	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
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Joshua C. Colp	3ed80fc57b	CHANGES: Change md file extension to txt. Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7 (cherry picked from commit `c9cd681261`)	6 years ago
Joshua C. Colp	21e9051461	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	82c3939c38	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
George Joseph	99efe1f868	Merge "codec negotiation: add incoming_call_offer_prefs option"	6 years ago
Joshua Colp	e8468eee13	Merge "res_rtp_asterisk: Add 'rtp show settings' cli command"	6 years ago
Rodrigo Ramírez Norambuena	e089779908	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
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Kevin Harwell	a715cf5aaa	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. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Kevin Harwell	999fdef335	Merge "app_mixmonitor: Turn on synchronization by default"	6 years ago
Sean Bright	8dcdce42a9	app_mixmonitor: Turn on synchronization by default The optional synchronization behavior created in `64906c4c9b` is now the default for MixMonitor. * Add a new flag 'n' that allows for this behavior to be turned off * Add a notice when the 'S' option is used indicating that it is no longer necessary Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4	6 years ago
Sean Bright	ddfb60ac2c	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	a72caa041f	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	b76ab5e5c9	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
George Joseph	9688381f05	Merge "http: Add ability to disable /httpstatus URI"	6 years ago
Friendly Automation	95c6fbeae0	Merge "app_voicemail: Remove MessageExists and MESSAGE_EXISTS()"	6 years ago
Sean Bright	0dce6f746b	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	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit `ec83b11183`) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit `fd64bb66f9`) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	6 years ago
Sean Bright	50d02d6194	pbx.c: Include filesystem cache in free memory calculation ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Friendly Automation	c665878e92	Merge "app_queue: Deprecate the QueueMemberPause.Reason field"	6 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	6 years ago
Sean Bright	0c2bf1664c	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	9522390a69	app_queue: Deprecate the QueueMemberPause.Reason field The QueueMemberPause AMI event includes two fields that return the reason a member was paused. * In release branches, deprecate Reason in favor of PausedReason. * In master, remove the Reason field entirely. ASTERISK-28349 #close Reported by: Niksa Baldun Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296	6 years ago
Friendly Automation	51f811183a	Merge "ARI: Ability to inhibit COLP frames when adding channels to a bridge"	6 years ago
Sean Bright	312abaa1fe	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
George Joseph	a4fd89536d	Merge "app_bridgeaddchan.c: Make BridgeAdd be more like Bridge"	6 years ago
George Joseph	6b7334a311	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail."	6 years ago
Friendly Automation	5b815fe1ac	Merge "app_dial.c: Simplify dialplan using Dial."	6 years ago
Richard Mudgett	fe3cce816c	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	19069f7db7	app_bridgeaddchan.c: Make BridgeAdd be more like Bridge * Made BridgeAdd not hangup the call if there is a problem. * Reduced message level from warning to verbose for normal exception cases. * Added a loop safety check to BridgeAdd. * Made BridgeAdd set BRIDGERESULT with the status when dialplan is resumed. Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426	6 years ago
Richard Mudgett	abcb4ab321	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	d86a6ac5ce	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
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	6 years ago
George Joseph	be93537382	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option"	6 years ago
Joshua C. Colp	89b7144fbd	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
Kevin Harwell	b6f5607359	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
Pascal Cadotte Michaud	bf4dd3d837	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	91cdb9537e	Merge "Revert "PJSIP_CONTACT: add missing argument documentation""	6 years ago
Joshua Colp	d0b198b330	Revert "PJSIP_CONTACT: add missing argument documentation" This reverts commit `7e3015d779`. 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: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f	6 years ago
Friendly Automation	2587b7e45f	Merge "PJSIP_CONTACT: add missing argument documentation"	6 years ago
George Joseph	7e3a6e158f	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	7e3015d779	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	d257a0898e	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	7362647e2f	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	b43cdc7f1e	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	b0b3bd627f	Merge "res_musiconhold: Add new 'playlist' mode"	6 years ago
Ben Ford	4c3655ecfd	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	966488ab52	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

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741 Commits (ef580f96e71be055d44dbe07ff63fe39ec697408)