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2008-01-09Set the caller id within the gtalk_alloc function.phsultan1-15/+4
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: http://svn.digium.com/svn/asterisk/branches/1.4@97489 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Don't try to allocate memory that we're just going to re-allocate later anyways.qwell1-3/+0
Issue 11130, patch by eliel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@87906 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Fix the output for this channel help CLI commandphsultan1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85800 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-13Closes issue #9401, reported and patched by irrot, with slightphsultan1-13/+29
modifications by me. Handle DTMF sent by Asterisk properly. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82309 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-06Various string length fixes. Removed an unused variable in aji_client ↵phsultan1-6/+6
structure (context) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81743 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-31Make the 'gtalk show channels' CLI command available.phsultan1-2/+43
Closes issue 10548, reported by keepitcool. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81410 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-24Closes issue #10509phsultan1-1/+27
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: http://svn.digium.com/svn/asterisk/branches/1.4@80661 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13(closes issue #10437)file1-2/+0
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: http://svn.digium.com/svn/asterisk/branches/1.4@79174 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Make payload IDs for iLBC/Speex match to our list. Since these are dynamic ↵file1-2/+2
payloads the other side shouldn't care. (issue #9426 reported by irroot) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@72331 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Don't modify a variable that we don't want modified. Make a copy of it instead.qwell1-7/+6
Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@72125 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Only attempt to queue a hangup on the owner channel if it actually exists.russell1-2/+2
(issue #9795, patch from zandbelt) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@70084 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-25handle the GNUTLS library properly in the configure script and build systemkpfleming1-0/+1
don't build in OSP support unless we have found and are allowed to use SSL support git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66157 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24oops, use #ifdef instead of #ifkpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65967 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24don't reference GnuTLS headers and functions unless the configure script ↵kpfleming1-1/+5
found it git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65966 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface ↵oej1-0/+6
did not support multithreading correctly. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65901 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!oej1-2/+21
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65892 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65857 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Issue #8536 - Caller ID not set in CDR for jingleoej1-3/+16
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65841 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09This is a big improvement over the current CDR fixes. It may still need ↵murf1-1/+1
refinement, but this won't have as many folks bothered. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60989 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Fix locking issue, and accept "transport-accept" as a valid accept message.qwell1-6/+9
This should solve issues 8970 and 8503. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55954 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Fix segfault when buddy couldn't be found.qwell1-1/+1
Issue 7764, patch by sailer git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55799 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-20No need to cast nor free with strdupa (thanks file)qwell1-10/+3
55555! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55555 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-09another dependencykpfleming1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53781 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-23Update channel drivers to use module referencing so that unloading them ↵file1-9/+2
while in use will not result in crashes. (issue #8897 reported by junky) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51788 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END arerussell1-3/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@51328 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merge the changes from the /team/group/vldtmf_fixup branch.russell1-6/+18
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: http://svn.digium.com/svn/asterisk/branches/1.4@51311 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Do not do a partial bridge for Google Talk since we need to handle STUN. ↵file1-4/+5
(issue #8448 reported by phsultan) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48168 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-08Make this module build againrussell1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47329 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07These mods are to solve the problem in bug 7506. It's a lot of rework to ↵murf1-8/+11
solve a fairly small problem... such is life. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47303 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-01bind address support from bug 8164mogorman1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46822 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-12fix for bug 7764.mogorman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44982 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-03fix issue with dialing client without resource.mogorman1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44312 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21updates for better compontent supportmogorman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43466 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18seperate jingle and gtalk so it will be easier to trackmogorman1-0/+1830
changes in both of the moving specs. Currently chan_gtalk is compatible with the latest gtalk/libjingle version, and chan_jingle needs a lot of work. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43185 f38db490-d61c-443f-a65b-d21fe96a405b
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