61 Commits (6353bddc57751ea2db044637fdd3f5b35b96f44c)

Author	SHA1	Message	Date
Tilghman Lesher	f491267c88	Merged revisions 114708 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	08e674bce0	Pass the hangup cause all the way to the calling app/channel. ... (closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	41fd7a6a21	(closes issue #6113) ... Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	71dc6a4771	Merged revisions 112820 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	db884798db	Merged revisions 112766 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	1c0bc928d1	Merged revisions 107714 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber). (closes issue #12014) Reported by: junky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	7293986e44	Remove unnecessary if statements before calling iks_delete (redundant check is ... done inside iks_delete), thus making the code conform with coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	55240a4e35	Merged revisions 97489 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	d2b29df4f0	Manager events from the "moremanager" branch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	7e8835e0d7	remove another set of redundant #include "asterisk/options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	0595b5e2aa	include "logger.h" and errno.h from asterisk.h - usage shows that they ... were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	d82a631f9c	more removal of duplicate #include lines ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7c56918262	Commit some cleanups to the format type code. ... - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	2c582c7cfb	Allow gtalk and jingle to use TLS connections again. ... Closes issue #9972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	2902601eea	Remove traces of gnutls, since we no longer use/need it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	fa33494d80	Merged revisions 87906 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11130) (closes issue #11132) ........ r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try to allocate memory that we're just going to re-allocate later anyways. Issues 11130 and 11132, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ebe4050128	Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b0f3e6097e	Convert NEW_CLI to AST_CLI. ... Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	65547b09b4	Fix CLI help output ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e97a723cf1	Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) ... (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	5734c0df49	Merged revisions 82309 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 Sep 2007) | 4 lines Closes issue #9401, reported and patched by irrot, with slight modifications by me. Handle DTMF sent by Asterisk properly. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	da620112de	Merged revisions 81743 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) | 1 line Various string length fixes. Removed an unused variable in aji_client structure (context) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	2fd2667d13	Merged revisions 81410 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command available. Closes issue 10548, reported by keepitcool. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	fce77d7d6a	Merged revisions 80661 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are answered too early, which results in CDRs wrongly stating that a call was ANSWERED when the calling party cancelled a call before before being established. We must not answer the call upon reception of a 'transport-accept' iq packet, but this packet still needs to be acknowledged, otherwise the remote peer would close the call (like in #8970). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	56b9568164	Don't reload a configuration file if nothing has changed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	d5eda8709c	Merged revisions 79174 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	22114b509d	Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	4897378ff1	Merged revisions 72331 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	8bf745194f	Merged revisions 72125 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines Don't modify a variable that we don't want modified. Make a copy of it instead. Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). Note: chan_jingle in trunk does not appear to have the same bug. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	3957ce9215	Merged revisions 70084 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines Only attempt to queue a hangup on the owner channel if it actually exists. (issue #9795, patch from zandbelt) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	055d82cbce	Add a massive set of changes for converting to use the ast_debug() macro. ... (issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	9d05ff8ed5	Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	ce9ec91897	ast_calloc janitor (Inspired by issue 9860) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	c74518e3ff	Merged revisions 66157 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines handle the GNUTLS library properly in the configure script and build system don't build in OSP support unless we have found and are allowed to use SSL support ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	287bcc6127	Merged revisions 65965-65967 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines don't use uninitialized variables ........ r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines don't reference GnuTLS headers and functions unless the configure script found it ........ r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use #ifdef instead of #if ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	b4f7d35240	Merged revisions 65901 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	bd2ae8b587	Merged revisions 65892 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	45bb9430b9	Merged revisions 65857 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	d3c7be548d	Merged revisions 65841 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines Issue #8536 - Caller ID not set in CDR for jingle ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	ecaf781933	Merged revisions 60989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b2ddaaf033	Add support for RTP packetization in chan_jingle and chan_gtalk. ... (issue #9416, phsultan) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	28a6129af8	Merged revisions 55954 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines Fix locking issue, and accept "transport-accept" as a valid accept message. This should solve issues 8970 and 8503. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8f28800765	Merged revisions 55799 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines Fix segfault when buddy couldn't be found. Issue 7764, patch by sailer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	ae47fc4541	Merged revisions 55555 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines No need to cast nor free with strdupa (thanks file) 55555! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ba32ee49d0	Adding Realtime Text support (T.140) to Asterisk ... T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	e758ef09b3	Merged revisions 53779-53781 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file ........ r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines add some inter-module dependencies ........ r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines another dependency ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	ee3ab150f6	Merged revisions 51788 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3275357f20	Merged revisions 51328 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same thing. So, a digit would have been interpreted incorrectly here. Since the channel driver will always have the begin and end callbacks called for a digit, only support the button-down and button-up messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dcca8f345f	Merged revisions 51311 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago