Olle Johansson
d2b29df4f0
Manager events from the "moremanager" branch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 19:50:12 +00:00
Luigi Rizzo
7e8835e0d7
remove another set of redundant #include "asterisk/options.h"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:24:55 +00:00
Luigi Rizzo
0595b5e2aa
include "logger.h" and errno.h from asterisk.h - usage shows that they
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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
2007-11-19 18:52:04 +00:00
Luigi Rizzo
d82a631f9c
more removal of duplicate #include lines
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 00:02:33 +00:00
Luigi Rizzo
fdb7f7ba3d
Start untangling header inclusion in a way that does not affect
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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
2007-11-16 20:04:58 +00:00
Tilghman Lesher
7c56918262
Commit some cleanups to the format type code.
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- 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
2007-11-06 22:51:48 +00:00
Jason Parker
2c582c7cfb
Allow gtalk and jingle to use TLS connections again.
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Closes issue #9972
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06 18:44:19 +00:00
Jason Parker
2902601eea
Remove traces of gnutls, since we no longer use/need it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01 23:26:51 +00:00
Jason Parker
fa33494d80
Merged revisions 87906 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11130 )
(closes issue #11132 )
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-31 21:18:52 +00:00
Jason Parker
ebe4050128
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 20:05:18 +00:00
Jason Parker
b0f3e6097e
Convert NEW_CLI to AST_CLI.
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Closes issue #11039 , as suggested by seanbright.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19 18:29:40 +00:00
Philippe Sultan
65547b09b4
Fix CLI help output
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16 10:38:57 +00:00
Russell Bryant
e97a723cf1
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
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(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
2007-10-11 19:03:06 +00:00
Philippe Sultan
5734c0df49
Merged revisions 82309 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13 11:54:56 +00:00
Philippe Sultan
da620112de
Merged revisions 81743 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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2007-09-06 17:00:58 +00:00
Philippe Sultan
2fd2667d13
Merged revisions 81410 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-31 17:43:50 +00:00
Philippe Sultan
fce77d7d6a
Merged revisions 80661 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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 ).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-24 11:49:36 +00:00
Tilghman Lesher
56b9568164
Don't reload a configuration file if nothing has changed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16 21:09:46 +00:00
Joshua Colp
d5eda8709c
Merged revisions 79174 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13 14:22:46 +00:00
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 21:44:58 +00:00
Joshua Colp
4897378ff1
Merged revisions 72331 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 22:58:53 +00:00
Jason Parker
8bf745194f
Merged revisions 72125 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 17:14:31 +00:00
Russell Bryant
3957ce9215
Merged revisions 70084 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19 19:15:03 +00:00
Russell Bryant
055d82cbce
Add a massive set of changes for converting to use the ast_debug() macro.
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(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
2007-06-14 19:39:12 +00:00
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Tilghman Lesher
ce9ec91897
ast_calloc janitor (Inspired by issue 9860)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-03 06:10:27 +00:00
Kevin P. Fleming
c74518e3ff
Merged revisions 66157 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-25 14:37:55 +00:00
Kevin P. Fleming
287bcc6127
Merged revisions 65965-65967 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines
don't use uninitialized variables
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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
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r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines
oops, use #ifdef instead of #if
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 19:05:42 +00:00
Olle Johansson
b4f7d35240
Merged revisions 65901 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 15:29:10 +00:00
Olle Johansson
bd2ae8b587
Merged revisions 65892 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 15:23:04 +00:00
Olle Johansson
45bb9430b9
Merged revisions 65857 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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2007-05-24 15:21:39 +00:00
Olle Johansson
d3c7be548d
Merged revisions 65841 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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2007-05-24 14:52:01 +00:00
Steve Murphy
ecaf781933
Merged revisions 60989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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;
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 05:41:34 +00:00
Russell Bryant
b2ddaaf033
Add support for RTP packetization in chan_jingle and chan_gtalk.
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(issue #9416 , phsultan)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-03 22:33:03 +00:00
Jason Parker
28a6129af8
Merged revisions 55954 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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2007-02-21 20:30:54 +00:00
Jason Parker
8f28800765
Merged revisions 55799 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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2007-02-21 02:04:10 +00:00
Jason Parker
ae47fc4541
Merged revisions 55555 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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!
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2007-02-20 16:56:58 +00:00
Olle Johansson
ba32ee49d0
Adding Realtime Text support (T.140) to Asterisk
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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
2007-02-16 13:35:44 +00:00
Kevin P. Fleming
e758ef09b3
Merged revisions 53779-53781 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines
add some inter-module dependencies
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r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines
another dependency
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2007-02-09 23:53:51 +00:00
Joshua Colp
ee3ab150f6
Merged revisions 51788 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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2007-01-23 22:59:55 +00:00
Russell Bryant
3275357f20
Merged revisions 51328 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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2007-01-19 19:09:04 +00:00
Russell Bryant
dcca8f345f
Merged revisions 51311 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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...)
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2007-01-19 18:06:03 +00:00
Russell Bryant
ec8591d04c
Constify a bunch of usage strings for CLI commands.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 07:35:31 +00:00
Joshua Colp
869101028b
Merged revisions 48168 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30 21:22:01 +00:00
Luigi Rizzo
c8597704ce
fix compilation.
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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
2006-11-08 07:21:45 +00:00
Steve Murphy
908f176cf3
A fair number of changes for the sake of bug 7506
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-07 21:47:49 +00:00
Luigi Rizzo
39d94767d7
remove useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:24:08 +00:00
Matt O'Gorman
09ef9b465b
Merged revisions 46822 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) | 2 lines
bind address support from bug 8164
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-01 20:38:05 +00:00
Matt O'Gorman
e20bb6fa69
Merged revisions 44982 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006) | 2 lines
fix for bug 7764.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-12 20:41:37 +00:00
Matt O'Gorman
7294ba3852
Merged revisions 44312 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r44312 | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2 lines
fix issue with dialing client without resource.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-03 22:36:51 +00:00