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${ noResults }
55 Commits (1036d0869d4b4036cd9f19f740f768fbd482c27e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
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
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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
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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
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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 |
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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 |
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Jason Parker
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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 |
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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 |
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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 |
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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 |
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Russell Bryant
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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
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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 |
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Russell Bryant
|
ec8591d04c |
Constify a bunch of usage strings for CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
|
869101028b |
Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Luigi Rizzo
|
c8597704ce |
fix compilation.
Overall i think the previous change to ast_channel_alloc() to close bug 7506 should have been done by defining an ast_set_callerid_noevent() function that does the setting without generating the event. Lot less code duplication, and easier to handle. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Steve Murphy
|
908f176cf3 |
A fair number of changes for the sake of bug 7506
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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39d94767d7 |
remove useless usecnt stuff
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Matt O'Gorman
|
09ef9b465b |
Merged revisions 46822 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.4 ........ r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) | 2 lines bind address support from bug 8164 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |