asterisk

Commit Graph

Author	SHA1	Message	Date
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	6 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	6 years ago
Asterisk Development Team	f1e0c9d1b8	Update CHANGES and UPGRADE.txt for 17.0.0	6 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	6 years ago
George Joseph	80d8dce6af	Prepare Asterisk 17 Branch Change-Id: Idb79a69646d2511e7bf1573b9b0322cc22ea54e8	6 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	6 years ago
George Joseph	93ccff25c6	Merge "app_attended_transfer: new application AttendedTransfer"	6 years ago
Friendly Automation	9bb9700e07	Merge "app_blind_transfer: new application BlindTransfer"	6 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(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
Alexei Gradinari	745cbab501	app_blind_transfer: new application BlindTransfer BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Kirsty Tyerman	bcaa01b024	pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi ASTERISK-28234 Reported-by: Kirsty Tyerman Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1	6 years ago
Friendly Automation	5750c97eb9	Merge "app_readexten: new option 'p' to stop reading on '#' key"	6 years ago
Ben Ford	9969c77bc2	build: Fix file format in CHANGES-staging. One of the change files doesn't conform to the format that the release scripts need in order to parse it. Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c	6 years ago
Alexei Gradinari	408210bd4c	app_readexten: new option 'p' to stop reading on '#' key This patch adds the 'p' option. The extension entered will be considered complete when a # is entered. Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1	6 years ago
George Joseph	be83591f99	res_rtp_asterisk: Add ability to propose local address in ICE You can now add the "include_local_address" flag to an entry in rtp.conf "[ice_host_candidates]" to include both the advertized address and the local address in ICE negotiation: [ice_host_candidates] 192.168.1.1 = 1.2.3.4,include_local_address This causes both 192.168.1.1 and 1.2.3.4 to be advertized. Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db	6 years ago
Joshua Colp	80dba268ea	app_confbridge: Add "all" variants of REMB behavior. When producing a combined REMB value the normal behavior is to have a REMB value which is unique for each sender based on all of their receivers. This can result in one sender having low bitrate while all the rest are high. This change adds "all" variants which produces a bridge level REMB value instead. All REMB reports are combined together into a single REMB value that is the same for each sender. ASTERISK-28401 Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c	6 years ago
Antoni Goldstein	8e21c25ce5	app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled at the earliest received PROGRESS or RINGING. Added millisecond versions of DIALEDTIME and ANSWEREDTIME. Added millisecond versions of ast_channel_get_up_time and ast_channel_get_duration in channel.c. ASTERISK-28363 Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1	6 years ago
Joshua Colp	9c070a4ae3	Merge "res_pjsip: Added a norefersub configuration setting"	6 years ago
Dan Cropp	cffa2a74cb	res_pjsip: Added a norefersub configuration setting Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	6 years ago
Ben Ford	a4ab7f5f80	build: Revise CHANGES and UPGRADE.txt handling. This changes the way that we handle adding changes to CHANGES and UPGRADE.txt. The reason for this is because whenever someone needed to make a change to one of these files and someone else had already done so, you would run into merge conflicts. With this new setup, there will never be merge conflicts since all changes will be documented in the doc/<file>-staging directory. The release script is now responsible for merging all of these changes into the appropriate files. There is a special format that these files have to follow in order to be parsed. The files do not need to have a meaningful name, but it is strongly recommended. For example, if you made a change to pjsip, you may have something like this "res_pjsip_relative_title", where "relative_title" is something more descriptive than that. Inside each file, you will need a subject line for your change, followed by a description. There can be multiple subject lines. The file may look something like this: Subject: res_pjsip Subject: Core A description that explains the changes made and why. The release script will handle the bulleting and section separators! You can still separate with new lines within your description. The headers ("Subject" and "Master-Only") are case sensative, but the value for "Master-Only" ("true" or "True") is not. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt ASTERISK-28111 #close Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47	6 years ago
Ben Ford	d5d8448ce5	build: Add staging directories for future changes. This is the first step in changing the release process so that changes made to the CHANGES and UPGRADE.txt files do not result in merge conflicts every time multiple people modify these files. The changes made will go in these new directories: doc/CHANGES-staging and doc/UPGRADE-staging. The README.md files explain how things will work, but here's a little overview. When you make a change that would go in either CHANGES or UPGRADE.txt, this should instead be documented in a new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory, respectively. The format will look like this: Subject: res_pjsip A description that explains the changes made and why. The release script will handle the bulleting and section separators! The 'Subject:' header is case-sensitive. You can still separate with new lines within your description. Subject: res_ari Master-Only: true You can have more than one subject, and they don't have to be the same! Also, the 'Master-Only' header should always be true and is also case-sensitive (but the value is not - you can have 'true' or 'True'). This header will only ever be present in the master branch. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt This is an initial change for ASTERISK_28111. Functionally, this will make no difference, but it will prep the directories for when the changes from CHANGES and UPGRADE.txt are extracted. Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6	6 years ago

31 Commits (d869882697fc3d4645a4b1599e1c967e3d22f70e)