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2008-07-11Merged revisions 130129 via svnmerge from bbryant1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@130130 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-02Merged revisions 119741 via svnmerge from phsultan1-8/+31
https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@119743 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Merged revisions 118644 via svnmerge from phsultan1-10/+42
https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 | phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 lines Changed to temporary namespaces to match with latest XEPs. As soon as Jingle is completely standardized, we can set those namespaces to their final values. Added two attributes to the jingle_pvt struct to store the content name attributes. Reported by Robert McQueen on Telepathy's framework mailing list : http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html Keeping working on our Jingle stack! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@118645 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Merged revisions 118614 via svnmerge from phsultan1-38/+40
https://origsvn.digium.com/svn/asterisk/trunk ........ r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) | 1 line Code simplification ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@118615 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Merged revisions 114709 via svnmerge from tilghman1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@114710 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107718 via svnmerge from qwell1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | 13 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@107720 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29Remove unnecessary if statements before calling iks_delete (redundant check isphsultan1-23/+21
done inside iks_delete), thus making the code conform with coding guidelines. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105263 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-17more removal of duplicate #include linesrizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89349 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Commit some cleanups to the format type code.tilghman1-5/+2
- 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: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Allow gtalk and jingle to use TLS connections again.qwell1-7/+7
Closes issue #9972 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89041 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Remove traces of gnutls, since we no longer use/need it.qwell1-10/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Merged revisions 87906 via svnmerge from qwell1-3/+1
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: http://svn.digium.com/svn/asterisk/trunk@87907 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-2/+2
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-2/+2
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Fix CLI help outputphsultan1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85787 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Added two CLI functions, taken from chan_gtalk :phsultan1-50/+107
- jingle reload ; - jingle show channels. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85778 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Make an audio path under the following call configuration :phsultan1-11/+18
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Modifications : - set bridge type to partial ; - process media candidates from the remote peer properly. Now we have Jingle audio, at least between two Asterisk Jingle clients. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85777 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-15Allow RTP structure registrationphsultan1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85555 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-09Remove redundant includes (patch by snuffy) (Closes issue #10922)tilghman1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-25Comply with latest XEP-0166, XEP-0167, XEP-0176.phsultan1-106/+157
No real Jingle implementation being available, testing was made using two Asterisk servers relaying SIP calls over their Jingle channels: SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was possible to test the code in both ways, and make the Jingle channel comply with the latest specifications. No sound available yet. Main modifications include : - modified the 'jingle_candidate' structure and the 'jingle_create_candidates' function according to XEP-0176 ; - modified the 'jingle_action' function in order to properly terminate a Jingle session, in conformance with XEP-0166 ; - modified username format used in STUN requests ; - actually make the bindaddr configuration field useable. Todo : - set audio paths up (no native bridging) ; - make the CLI gtalk functions available to jingle ; - clean up the storage space used in strings. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83743 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-19Replace Google namespace occurrences with Jingle. The former namespacephsultan1-22/+22
is handled by chan_gtalk. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83076 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-19Remove namespaces in payload-type tags.phsultan1-11/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83072 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-19Transmit proper invitation, thus conforming to XEP-0166 (Jingle generalphsultan1-19/+33
specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 (Jingle ICE Transport). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83055 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-14Fix DTMF following what has been done in issue #9401. Thanks irroot.phsultan1-5/+22
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82373 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-13Modify rule filters to match with the Jingle namespace constantphsultan1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82320 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-13Changed Jingle and Jingle DTMF namespaces.phsultan1-6/+6
As both specifications are in the Experimental status, the namespaces specified therein shall be of the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See the Namespace issuance section in XEP-0053 : http://www.xmpp.org/extensions/xep-0053.html#namespaces git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-13Reflect Jingle DTMF specification changesphsultan1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82312 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Merged revisions 79174 via svnmerge from file1-2/+0
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: http://svn.digium.com/svn/asterisk/trunk@79175 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Add support for using epoll instead of poll. This should increase ↵file1-4/+4
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Silly jingle...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72358 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Merged revisions 70084 via svnmerge from russell1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@70088 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.russell1-16/+9
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-03ast_calloc janitor (Inspired by issue 9860)tilghman1-10/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66981 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-25more minor fixeskpfleming1-0/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66175 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-25Merged revisions 66157 via svnmerge from kpfleming1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@66158 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Adding external referenses for doxygenoej1-0/+2
See http://www.asterisk.org/doxygen/trunk/extref.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63230 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10updated ast_channel_alloc() call to include the 4 extra args everyone got. ↵murf1-1/+1
Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-03Add support for RTP packetization in chan_jingle and chan_gtalk.russell1-0/+8
(issue #9416, phsultan) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60011 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Merged revisions 55954 via svnmerge from qwell1-3/+6
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: http://svn.digium.com/svn/asterisk/trunk@55955 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Merged revisions 55799 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@55805 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-20Merged revisions 55555 via svnmerge from qwell1-5/+1
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: http://svn.digium.com/svn/asterisk/trunk@55556 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-17Update chan_jingle to new definition of set_rtp_peer.file1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@55088 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10add another dependencyrussell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53785 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merged revisions 51311 via svnmerge from russell1-11/+13
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: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-08fix compilation.rizzo1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@47306 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07A fair number of changes for the sake of bug 7506murf1-8/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47290 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-13 10:49:28 +0000