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2010-03-02Merged revisions 249893 via svnmerge from dvossel1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249895 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-1/+2
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-09Merge more changes from the resolve-shadow-warnings branch (henceforth knownseanbright1-3/+3
as RSW since i am too lazy to keep typing it all out). This time a few of the channels. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136888 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Janitor patch to change uses of sizeof to ARRAY_LENbbryant1-2/+2
(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/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-02Do not link the guest account with any configured XMPP client (inphsultan1-8/+31
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/trunk@119741 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Changed to temporary namespaces to match with latest XEPs. As soon asphsultan1-10/+42
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/trunk@118644 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Code simplificationphsultan1-38/+40
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118614 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak1-1/+1
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Merged revisions 114708 via svnmerge from tilghman1-2/+7
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/trunk@114709 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-24Pass the hangup cause all the way to the calling app/channel.mvanbaak1-1/+1
(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: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107714 via svnmerge from qwell1-0/+7
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/trunk@107718 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
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