asterisk

Commit Graph

Author	SHA1	Message	Date
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
George Joseph	799c4cf494	Merge "chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS"	6 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	6 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	6 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	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
Corey Farrell	879e592baf	Build System: Enable python3 compatibility. * Consistently use spaces in rest-api-templates/asterisk_processor.py. * Exclude third-party from docs/full-en_US.xml. * Add docs/full-en_US.xml to .gitignore. * Use list() to convert python3 view. * Use python3 print function. * Replace cmp() with equivalent equation. * Replace reference to out of scope subtype variable with name parameter. * Use unescaping triple bracket notation in mustache templates where needed. This causes behavior of Python2 to be maintained when using Python3. * Fix references to has_websocket / is_websocket in res_ari_resource.c.mustache. * Update calculation of has_websocket to use any(). * Use unicode mode for writing output file in transform.py. * Replace 'from swagger_model import *' with explicit import of required symbols. I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the print syntax has been fixed. Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33	7 years ago
Corey Farrell	0ad13949c1	doc/lang/language-criteria.txt: Link to wiki. This document is out of date and is superseded by content on the Asterisk wiki. ASTERISK-24386 #close Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822	7 years ago
Sean Bright	fc98843d4b	appdocsxml.xslt: Add Language to channel snapshot transformation Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8	7 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Sean Bright	9415ec2877	docs: Remove old API changes documentation Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a	8 years ago
Corey Farrell	9316a064fd	README: Send people to secure websites where available. We should be sending people to secure web URL's where available. Update README's and docs. Change-Id: Id5b1e049b0b18b49a784f1254605aefa244ce19a	8 years ago
Matt Jordan	a72ef38113	res/ari/resource_bridges: Add the ability to manipulate the video source In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	9 years ago
Corey Farrell	824a4e84d1	Refactor usage pattern of xmldoc info tag. This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	9 years ago
Matt Jordan	ddab42e296	func_channel: Reorganize documentation * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b	9 years ago
George Joseph	1d3191b118	progdocs: Exclude ./third-party from documentation generation We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17	9 years ago
Richard Mudgett	c2d48a2a28	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	10 years ago
Corey Farrell	57386dcb67	Allow command-line options to override asterisk.conf. Previous versions of Asterisk processed command-line options before processing asterisk.conf. This meant that if an option was set in asterisk.conf, it could not be overridden with the equivelent command line option. This change causes Asterisk to process the command-line twice. First it processes options that are needed to load asterisk.conf, then it processes the remaining options after the config is read. This changes the function of -X slightly. Previously using -X without disabling execincludes in asterisk.conf caused #exec to be usable in any config. Now -X only enables #exec for the load of asterisk.conf, if it is wanted in the rest of the system it must be enabled with execincludes in asterisk.conf. Updated 'asterisk -h' and 'man asterisk' to reflect the limited function of -X. ASTERISK-25042 #close Reported by: Corey Farrell Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7	10 years ago
George Joseph	cf637f2510	doc: Make progdocs play nice with git Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in Changed /Makefile to copy asterisk-ng-doxygen.in to asterisk-ng-doxygen then modify it with version instead of modifying asterisk-ng-doxygen directly. Updated clean targets as well. Updated /.gitignore and doc/.gitignore. Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622	10 years ago
Corey Farrell	c232ff3af0	Git Migration: Create doc/rest-api when needed. Create the directory './doc/rest-api' at the start of 'make ari-stubs' to prevent an error when documentation is generated. The directory is also added to git ignores. ASTERISK-25027 Reported by: Corey Farrell Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b	10 years ago
George Joseph	b35e184d41	Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Tested-by: George Joseph	10 years ago
Walter Doekes	b8c1130ed1	docs: Escape unescaped minus sign in asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561) ........ Merged revisions 423915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423916 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423917 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423918 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	664f83a03b	doc: Add a manpage for the smsq utility This patch adds a manpage for the smsq utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422376 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422377 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422378 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422379 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	81598fa082	doc: Add a manpage for the aelparse utility This patch adds a manpage for the aelparse utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422372 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422373 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422374 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	922e3203a9	xmldocs: Add support for an <example> tag in the Asterisk XML Documentation This patch adds support for an <example /> tag in the XML documentation schema. For CLI help, this doesn't change the formatting too much: - Preceeding white space is removed - Unlike with para elements, new lines are preserved However, having an <example /> tag in the XML schema allows for the wiki documentation generation script to surround the documentation with {code} or {noformat} tags, generating much better content for the wiki - and allowing us to put dialplan examples (and other code snippets, if desired) into the documentation for an application/function/AMI command/etc. Review: https://reviewboard.asterisk.org/r/3807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4445ee7fc0	AMI: Allow for command response documentation Allow for responses to AMI actions/commands to be documented properly in XML and displayed via the CLI. Response events are documented exactly as standard AMI events are documented. Review: https://reviewboard.asterisk.org/r/3812/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b0bb03e916	bridging: Give bridges a name and a known creator Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ ........ Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f46b30bd36	func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint details This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint, any property configured on an endpoint. This function is a companion to the CHANNEL function, which can be used to extract the endpoint name for a channel. Review: https://reviewboard.asterisk.org/r/3035 ........ Merged revisions 403616 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	e2204a1e61	man pages for astdb2bdb and astdb2sqlite3 Review: https://reviewboard.asterisk.org/r/2898/ ........ Merged revisions 400279 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400281 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	4710869133	Add documentation for features configuration. Review: https://reviewboard.asterisk.org/r/2616 (closes issue ASTERISK-21542) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d04ab3c645	Add CLI configuration documentation This patch allows a module to define its configuration in XML in source, such that it can be parsed by the XML documentation engine. Documentation is generated in a two-pass approach: 1. The documentation is first generated from the XML pulled from the source 2. The documentation is then enhanced by the registration of configuration options that use the configuration framework This patch include configuration documentation for the following modules: * chan_motif * res_xmpp * app_confbridge * app_skel * udptl Two new CLI commands have been added: * config show help - show configuration help by module, category, and item * xmldoc dump - dump the in-memory representation of the XML documentation to a new XML file. Review: https://reviewboard.asterisk.org/r/2278 Review: https://reviewboard.asterisk.org/r/2058 patches: on review 2058 uploaded by twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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685 Commits (87f8ca2812b45e2c1eb8f5897b35575d2ded4e23)