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2011-02-08Documentation Updateslathama2-234/+958
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: http://svn.digium.com/svn/asterisk/trunk@307041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Documentation Updates.lathama1-4/+4
Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306827 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294745 via svnmerge from russell1-0/+0
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: http://svn.digium.com/svn/asterisk/trunk@294749 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294740 via svnmerge from russell92-20741/+10
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: http://svn.digium.com/svn/asterisk/trunk@294741 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Merged revisions 291725 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@291726 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288082 via svnmerge from rmudgett1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@288083 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286647 via svnmerge from rmudgett2-24/+24
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: http://svn.digium.com/svn/asterisk/trunk@286648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 285992 via svnmerge from diruggles1-4/+10
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: http://svn.digium.com/svn/asterisk/trunk@285993 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284698 via svnmerge fromrmudgett3-5/+346
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: http://svn.digium.com/svn/asterisk/trunk@284699 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-16Merged revisions 282470 via svnmerge from lmadsen2-0/+83
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: http://svn.digium.com/svn/asterisk/trunk@282471 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Merged revisions 280740 via svnmerge from tilghman3-141/+229
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: http://svn.digium.com/svn/asterisk/trunk@280741 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Some left-over hyphen-minus fixes in the man pagetzafrir1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278947 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inoej1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@276951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.tringenbach2-8/+12
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: http://svn.digium.com/svn/asterisk/trunk@276349 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Changed OSP TCP port from 1080 to 5045.transnexus1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274492 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Use the underscore package so that underscores do not need to be escaped.russell5-43/+44
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272684 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-25Merged revisions 272562 via svnmerge from tilghman1-5/+5
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: http://svn.digium.com/svn/asterisk/trunk@272568 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Update formatting for channelvariables.texpabelanger1-3/+3
(closes issue #17511) Reported by: klaus3000 Patches: channelvariables.tex-patch.txt uploaded by klaus3000 (license 65) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270801 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Add distributed devicestate via the XMPP protocol.tilghman1-0/+433
(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: http://svn.digium.com/svn/asterisk/trunk@270519 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-12Merged revisions 270078 via svnmerge from pabelanger1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@270079 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Add documentation explaining PLC in Asterisk.mmichelson2-0/+142
Review: https://reviewboard.asterisk.org/r/688/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269749 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add SRTP support for Asterisktwilson2-0/+48
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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Update CHANGES and aoc help doc to reflect AOC additionsdvossel1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267181 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett1-0/+180
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: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Merge the rest of the FullyBooted patchtwilson1-0/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265467 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Add the FullyBooted AMI eventtwilson1-2/+15
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: http://svn.digium.com/svn/asterisk/trunk@265320 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Merged revisions 260569 via svnmerge from lmadsen1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@260570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Asterisk data retrieval API.eliel1-0/+10
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: http://svn.digium.com/svn/asterisk/trunk@258517 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Add MEETMEBOOKID from r256019.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258515 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Missed this when reverting the bad version change in asterisk.tex.lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258387 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Fix change in asterisk.tex that got merged in after testing.lmadsen1-1/+1
(issue #17220) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258383 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Add ability to generate ASCII documentation from the TeX files.lmadsen8-15/+47
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: http://svn.digium.com/svn/asterisk/trunk@258351 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21fix whitespace issuejmls1-10/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258256 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added NEW ACTIONS entry for new MixMonitorMute AMI command.jmls2-1/+13
Added State and Direction variables for new MixMonitorMute AMI command. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258228 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257426 via svnmerge from lmadsen1-27/+105
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: http://svn.digium.com/svn/asterisk/trunk@257427 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257342 via svnmerge from lmadsen1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@257343 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-12Merged revisions 256900 via svnmerge from lmadsen1-0/+89
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: http://svn.digium.com/svn/asterisk/trunk@256901 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-10fix hyphen vs. minus in man pagestzafrir1-47/+47
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: http://svn.digium.com/svn/asterisk/trunk@256704 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge CCSS architecture document from CCSS branch.rmudgett1-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256608 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson2-0/+417
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: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-05Fix for localchannel.tex to allow PDFs to be generated again.lmadsen1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256161 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Update to new Local channel documentation.lmadsen1-28/+46
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: http://svn.digium.com/svn/asterisk/trunk@253256 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Update existing Local channel documentation.lmadsen1-57/+461
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: http://svn.digium.com/svn/asterisk/trunk@250609 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update IMAP documentation.lmadsen1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@250051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Update documentation to not imply we support overriding options.lmadsen1-12/+20
(closes issue #16855) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250037 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Enable SendText to send strings in encoded format.tilghman1-0/+2
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241364 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18Updated ExternalIVR documentationdiruggles1-61/+99
Rewrote a large portion of the existing documentation and added information about the TCP/IP socket interface git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240973 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-14Add documentation about how to build queues.lmadsen1-0/+823
Add a how-to set of documentation about building queues with Asterisk. This documentation is based on Asterisk 1.6.2 but should work on most versions with minor modifications. (closes issue #16237) Reported by: lmadsen Patches: Building Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by: pdhales, lmadsen, cmdrwalrus git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240039 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13Updated channel variable list of osplookup application.transnexus1-9/+42
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239625 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-30Add app_voicemail and say.c support for Vietnamese.qwell1-0/+0
Also add an XXX comment that I'm baffled nobody has ever complained about. We say "first message", and then we go into language-specific stuff where we proceed to say..."first message". (closes issue #15053) Reported by: dinhtrung Patches: vietnamese.ods uploaded by dinhtrung (license 776) app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded by dinhtrung (license 776) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237050 f38db490-d61c-443f-a65b-d21fe96a405b
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