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${ noResults }
628 Commits (ac6ec71fd23d4fe0ce9a7bdd43d9461b4732c713)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Matthew Nicholson
|
bb07ca66a1 |
Merged revisions 340109 via svnmerge from
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 |
14 years ago |
Jason Parker
|
5fe104ddb9 |
Merged revisions 331143 via svnmerge from
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 |
14 years ago |
|
Jason Parker
|
873962f772 |
Merged revisions 331139 via svnmerge from
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 |
14 years ago |
|
Jason Parker
|
19c8278815 |
Merged revisions 331138 via svnmerge from
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 |
14 years ago |
Andrew Latham
|
a350924700 |
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@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Andrew Latham
|
c4271440ab |
Documentation Updates.
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 |
14 years ago |
Russell Bryant
|
b1aa9069fe |
Merged revisions 294745 via svnmerge from
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 |
15 years ago |
|
Russell Bryant
|
893ca656af |
Merged revisions 294740 via svnmerge from
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 |
15 years ago |
|
Russell Bryant
|
6b1019418a |
Merged revisions 291725 via svnmerge from
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 |
15 years ago |
Richard Mudgett
|
e100d8f649 |
Merged revisions 288082 via svnmerge from
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 |
15 years ago |
|
Richard Mudgett
|
5fa454010e |
Merged revisions 286647 via svnmerge from
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 |
15 years ago |
David Ruggles
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c142342f12 |
Merged revisions 285992 via svnmerge from
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 |
15 years ago |
|
Richard Mudgett
|
df6d017aa1 |
Merged revisions 284698 via svnmerge from
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 |
15 years ago |
Leif Madsen
|
fc55d25510 |
Merged revisions 282470 via svnmerge from
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 |
15 years ago |
Tilghman Lesher
|
ee6592684b |
Merged revisions 280740 via svnmerge from
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 |
15 years ago |
Tzafrir Cohen
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76c1992f38 |
Some left-over hyphen-minus fixes in the man page
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Olle Johansson
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e129b31fc6 |
Add ability to configure the Max-Forwards header in the dialplan, as well as in
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 |
15 years ago |
Tim Ringenbach
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e19a6c248f |
Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.
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 |
15 years ago |
TransNexus OSP Development
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d6a70b619c |
Changed OSP TCP port from 1080 to 5045.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274492 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Russell Bryant
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910cd8be1d |
Use the underscore package so that underscores do not need to be escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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7aa382b53f |
Merged revisions 272562 via svnmerge from
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 |
15 years ago |
Paul Belanger
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5d2bbe86ad |
Update formatting for channelvariables.tex
(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 |
15 years ago |
|
Tilghman Lesher
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81c15adfa2 |
Add distributed devicestate via the XMPP protocol.
(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 |
15 years ago |
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Paul Belanger
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f9bcb15db3 |
Merged revisions 270078 via svnmerge from
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 |
15 years ago |
Mark Michelson
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326c783685 |
Add documentation explaining PLC in Asterisk.
Review: https://reviewboard.asterisk.org/r/688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Terry Wilson
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857814f435 |
Add SRTP support for Asterisk
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 |
15 years ago |
David Vossel
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3280a5c0af |
Update CHANGES and aoc help doc to reflect AOC additions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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afd4454c44 |
Generic Advice of Charge.
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 |
15 years ago |
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Terry Wilson
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0390dae08d |
Merge the rest of the FullyBooted patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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3c9a8ebadb |
Add the FullyBooted AMI event
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 |
15 years ago |
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Leif Madsen
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3a1cabb13f |
Merged revisions 260569 via svnmerge from
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 |
15 years ago |
Eliel C. Sardanons
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a753e8878b |
Asterisk data retrieval API.
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 |
15 years ago |
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Russell Bryant
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8096f0fecc |
Add MEETMEBOOKID from r256019.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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5258e5e683 |
Missed this when reverting the bad version change in asterisk.tex.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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8ea4ecd58a |
Fix change in asterisk.tex that got merged in after testing.
(issue #17220) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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8b11ae2e4f |
Add ability to generate ASCII documentation from the TeX files.
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 |
15 years ago |
Julian Lyndon-Smith
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4b7c56ccef |
fix whitespace issue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Julian Lyndon-Smith
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81fd235286 |
Added NEW ACTIONS entry for new MixMonitorMute AMI command.
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 |
15 years ago |
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Leif Madsen
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bef07ecc16 |
Merged revisions 257426 via svnmerge from
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 |
15 years ago |
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Leif Madsen
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28b78c5e67 |
Merged revisions 257342 via svnmerge from
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 |
15 years ago |
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Leif Madsen
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47d8fdae97 |
Merged revisions 256900 via svnmerge from
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 |
15 years ago |
|
Tzafrir Cohen
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16774318f9 |
fix hyphen vs. minus in man pages
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 |
15 years ago |
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Richard Mudgett
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d66b44b4ca |
Merge CCSS architecture document from CCSS branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Mark Michelson
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e24661fd18 |
Merge Call completion support into trunk.
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 |
15 years ago |
|
Leif Madsen
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3a4baef6d9 |
Fix for localchannel.tex to allow PDFs to be generated again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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b6e783d97c |
Update to new Local channel documentation.
Add same changes as commit to 1.4, but convert to TeX. (issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
|
06ff72a78e |
Update existing Local channel documentation.
A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
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491ea82b6d |
Update IMAP documentation.
Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Leif Madsen
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aec5c6f840 |
Update documentation to not imply we support overriding options.
(closes issue #16855) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250037 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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873989db91 |
Enable SendText to send strings in encoded format.
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |