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2012-09-22Doxygen Updates Janitor WorkAndrew Latham
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30Restore CODING-GUIDELINES to doc folderMatthew Jordan
In r294740, the CODING-GUIDELINES was removed from the doc folder in favor of the content on the Asterisk wiki. Some folks still look in the doc folder initially for coding guideline suggestions; as such, this patch adds a CODING-GUIDELINES file back into the doc folder. The content of the file merely points to the correct page on the Asterisk wiki where the coding guidelines currently live. (closes issue ASTERISK-20279) Reported by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 371961 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371962 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371963 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Add the ability to specify technology specific documentationMatthew Jordan
A number of applications/AMI commands in Asterisk have specific behavioral differences depending on the resource or channel technology those applications are executed on. For example, the MessageSend application/ command is technology agnostic, but how the channel drivers that support that functionality behave is dependant on the protocols and channel driver implementation. Prior to this patch, those details were either documented in the application/command documentation itself, or were left undocumented. This patch adds a new element to the documentation schema, <info/>. An info node is essentially a piece of technology specific reference information that can be included by any top level XML documentation node. For example, the MessageSend application can now include XMPP/SIP specific information, where that technology specific information can be defined in chan_motif/res_xmpp/ chan_sip. Likewise, that information can also be included in the MessageSend AMI command. Review: https://reviewboard.asterisk.org/r/2049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Add some additional documentation for core AMI eventsMatthew Jordan
This patch adds some basic documentation for a number of modules. This includes core source files in Asterisk (those in main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD has also been updated to allow referencing of AMI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Add AMI event documentationMatthew Jordan
This patch adds the core changes necessary to support AMI event documentation in the source files of Asterisk, and adds documentation to those AMI events defined in the core application modules. Event documentation is built from the source by two new python scripts, located in build_tools: get_documentation.py and post_process_documentation.py. The get_documentation.py script mirrors the actions of the existing AWK get_documentation scripts, except that it will scan the entirety of a source file for Asterisk documentation. Upon encountering it, if the documentation happens to be an AMI event, it will attempt to extract information about the event directly from the manager event macro calls that raise the event. The post_process_documentation.py script combines manager event instances that are the same event but documented in multiple source files. It generates the final core-[lang].xml file. As this process can take longer to complete than a typical 'make all', it is only performed if a new make target, 'full', is chosen. Review: https://reviewboard.asterisk.org/r/1967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-10Merged revisions 340109 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Merged revisions 331143 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line Regenerate asterisk man page from sgml. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Merged revisions 331139 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Merged revisions 331138 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line Revert merge of r306999, due to merge conflict. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Documentation UpdatesAndrew Latham
Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Documentation Updates.Andrew Latham
Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-11Merged revisions 294745 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) | 6 lines Remove CCSS architecture PDF. It has been moved to: https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-11Merged revisions 294740 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines Remove most of the contents of the doc dir in favor of the wiki content. This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-14Merged revisions 291725 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 Oct 2010) | 2 lines Fix a typo - s/seucre/secure/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 288082 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 Sep 2010) | 1 line Add note in party manipulation chapter on interception macros. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-14Merged revisions 286647 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) | 1 line Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-10Merged revisions 285992 via svnmerge from David Ruggles
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line Added missing documentation for ExternalIVR feature added in January 2010 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284698 via svnmerge fromRichard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines Added documentation for CONNECTEDLINE and REDIRECTING functions. (closes issue #17808) Reported by: jtodd Review: https://reviewboard.asterisk.org/r/875/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-16Merged revisions 282470 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines Merged revisions 282469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines Add information about creating sounds files using the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Merged revisions 280740 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines Merged revisions 280739 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines Document -B and -W flags and regenerate manpage from sgml ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23Some left-over hyphen-minus fixes in the man pageTzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inOlle Johansson
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.Tim Ringenbach
Change the documented pgsql schema to use "timestamp" instead of "time", as the latter is only a time without a date. Added some missing columns for cel's pgsql schema, and corrected spelling on some others. Updated cel's uniqueid size to be the same as the cdr. Added id column to cel's pgsql schema and updated code to allow unknown columns to get their default value instead of forcing 0 or empty string. Added microseconds to the timestamp cel logs to pgsql. Review: https://reviewboard.asterisk.org/r/734 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-07Changed OSP TCP port from 1080 to 5045.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-28Use the underscore package so that underscores do not need to be escaped.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-25Merged revisions 272562 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines Make the structure of the table specified before match the queries and results. (closes issue #17557) Reported by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16Update formatting for channelvariables.texPaul Belanger
(closes issue #17511) Reported by: klaus3000 Patches: channelvariables.tex-patch.txt uploaded by klaus3000 (license 65) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Add distributed devicestate via the XMPP protocol.Tilghman Lesher
(closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-12Merged revisions 270078 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines Fix typo in example ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-10Add documentation explaining PLC in Asterisk.Mark Michelson
Review: https://reviewboard.asterisk.org/r/688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Add SRTP support for AsteriskTerry Wilson
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Update CHANGES and aoc help doc to reflect AOC additionsDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Generic Advice of Charge.Richard Mudgett
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24Merge the rest of the FullyBooted patchTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24Add the FullyBooted AMI eventTerry Wilson
It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-03Merged revisions 260569 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22Asterisk data retrieval API.Eliel C. Sardanons
This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22Add MEETMEBOOKID from r256019.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21Missed this when reverting the bad version change in asterisk.tex.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21Fix change in asterisk.tex that got merged in after testing.Leif Madsen
(issue #17220) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21Add ability to generate ASCII documentation from the TeX files.Leif Madsen
These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21fix whitespace issueJulian Lyndon-Smith
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21Added NEW ACTIONS entry for new MixMonitorMute AMI command.Julian Lyndon-Smith
Added State and Direction variables for new MixMonitorMute AMI command. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-15Merged revisions 257426 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-15Merged revisions 257342 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-12Merged revisions 256900 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines Add How-To document on collecting debugging info for issues.asterisk.org Paul Belanger has been helping a lot with bug tracking recently and created this document that we can now point to when additional debugging information is required. This document will help those filing issues to know how to get the information required when filing their issues. This will make things easier on the developers. Initial text and changes by pabelanger. Tweaks and editing by myself. (closes issue #17159) Reported by: pabelanger Patches: HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-10fix hyphen vs. minus in man pagesTzafrir Cohen
In troff '-' is used for a hyphen. A minus is denoted by '\-' . This is normally also used for a dash. This patch converts all '-'-s that are minuses or dashes to '\-'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09Merge CCSS architecture document from CCSS branch.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09Merge Call completion support into trunk.Mark Michelson
From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-05Fix for localchannel.tex to allow PDFs to be generated again.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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