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2007-09-21gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set ofrussell1-2/+2
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, patch from qwell) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@83432 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
2007-07-18Don't bother reloading chan_h323 if it did not load successfully in the ↵file1-0/+4
first place. This would otherwise cause a crash. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@75619 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25Build a peer as well when hash323 is enabled in users.conf (issue #9599 ↵file1-0/+5
reported by asagage) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@71576 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
2007-02-10Fix VLDTMF receptionpcadach1-4/+18
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53881 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01chan_h323 is very stable, so let it built by defaultpcadach1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53057 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-31Merged revisions 53045 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines Fix a bunch of places where pthread_attr_init() was called, but pthread_attr_destroy() was not. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53046 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-30Revert reprecation of h.323 gk cycle command from pre-1.4 version instead of ↵pcadach1-1/+1
duplicated h323 cycle gk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52809 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-23Do not abort Asterisk startup if h323 configuration file not found (reported ↵pcadach1-3/+9
by mithraen) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51615 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merge the changes from the /team/group/vldtmf_fixup branch.russell1-2/+2
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
2006-11-10Let's give this a go...file1-11/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47457 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-10Discussion of these CLI changes resulted in more consistency (Bug 8236)tilghman1-17/+32
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47436 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-10Fix building of chan_h323 by completeing some structure definitions. (issue ↵file1-4/+6
#8327 reported by Mithraen) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47405 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07These mods are to solve the problem in bug 7506. It's a lot of rework to ↵murf1-15/+17
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
2006-11-02More changes making the CLI more consistent with "category verb arguments" ↵tilghman1-6/+16
(continuation of issue 8236) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47053 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25apparently developers are still not aware that they should be use ↵kpfleming1-43/+42
ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46200 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25add passthrough and file format support for G.722 16KHz audio (issue #5084, ↵kpfleming1-0/+2
original patch by andrew, updated by mithraen) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46154 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-07Propagate caller's transfer capability toopcadach1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44684 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04update thread creation code a bitkpfleming1-1/+1
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44378 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-02Optimization of oh323_indicate(): less locks - less problems, plus single ↵pcadach1-36/+17
exit point git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44166 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-01Do not simulate any audio tones if we got PROGRESS messagepcadach1-2/+11
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44135 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-29Pass TON/PRESENTATION information toopcadach1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44009 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-25Backport changes of trunk:pcadach1-19/+21
1) r43540: Avoid possible deadlock on channel destruction 2) r43590: Disable fastStart if requested by remote side git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43626 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-25Merged revisions 43472,43495 from trunkpcadach1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43582 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Remove unnecessary (long time ago commented out) codepcadach1-34/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43350 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Merge in latest round of chan_h323 changes. These are all isolated to ↵file1-3/+24
chan_h323 so meh. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43331 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-19Mergeing in Paul Cadach's chan_h323 changes *holds breath*mattf1-755/+1459
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43281 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-37/+39
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31everything that loads a config that needs a config file to runmogorman1-1/+1
now reports AST_MODULE_LOAD_DECLINE when loading if config file is not there, also fixed an error in res_config_pgsql where it had a non static function when it should. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41633 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵file1-3/+11
Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-12/+5
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-16move the calls to ast_jb_configure() to before the PBX thread is started on therussell1-4/+2
channel to remove the theoretical race condition that the channel could get bridged before the channel's jitterbuffer gets configured. This was pointed out by PCadach on IRC. Thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39964 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-15complete the coding style changes for these frame structureskpfleming1-6/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39832 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-15Made "style" change requested by Kevin Fleming... used initializer in ↵murf1-17/+3
declaration, brought decl down to block where the variable is used, got rid of memset. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39806 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-15On behalf of PCadach, " IMHO that was last deadlock in chan_h323..."murf1-13/+36
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39779 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-05Merged revisions 38903-38904 via svnmerge from russell1-11/+12
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines suppress a compiler warning about the usage of a potentially uninitialized variable ........ r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines Fix an issue that would cause a NewCallerID manager event to be generated before the channel's NewChannel event. This was due to a somewhat recent change that included using ast_set_callerid() where it wasn't before. This function should not be used in the channel driver "new" functions. (issue #7654, fixed by me) Also, fix a couple minor bugs in usecount handling. chan_iax2 could have increased the usecount but then returned an error. The place where chan_sip increased the usecount did not call ast_update_usecount() ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38905 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵kpfleming1-0/+10
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-03Blocked revisions 36725 via svnmergerussell1-10/+9
........ r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines use ast_set_callerid to be more consistent and to make sure that the "callerid" option in the conf files is always handled the same way and sets ANI (issue #7285, gkloepfer) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36726 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-23revert my changes that converted the jb on the channel to be dynamicallyrussell1-13/+8
allocated. These changes caused crashes when using a channel type that did not support the jitterbuffer. Instead of fixing why it's crashing, I'm going to implement this in a better way next week. The way I did it caused a jitterbuffer to be allocated on every channel where the channel type supported jitterbuffers, even if they were disabled. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35746 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-22- dynamically allocate the ast_jb structure that is on the channel structurerussell1-8/+13
so that channels not using a jitterbuffer don't waste as much memory - ensure that the channel drivers that use jitterbuffers can handle a failure from configuring a jitterbuffer on a new channel because of a memory allocation error - On passing through these channel drivers, configure the jitterbuffer before starting the PBX thread instead of afterwards. If the pbx fails to start for whatever reason, this would have caused a crash. - Also on passing, move the increase of the usecount to after all of the possible failure conditions in the function - fix a place where ast_update_use_count() was not called - ensure that the owner channel pointer of the channel pvt strcutures is set to NULL in failure conditions git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35553 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-4/+15
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-01- add the ability to configure forced jitterbuffers on h323, jingle,russell1-1/+26
and mgcp channels - remove the jitterbuffer configuration from the pvt structures in the sip, zap, and skinny channel drivers, as copying the same global configuration into each pvt structure has no benefit. - update and fix some typos in jitterbuffer related documentation (issue #7257, north, with additional updates and modifications) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31413 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-31update the rest of the channel drivers that use RTP so that their channelrussell1-1/+1
tech structures indicate that they create jitter git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31077 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10update the ->indicate() callback for the new argumentskpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26495 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-29partial adaptation to the new module loadingrizzo1-24/+9
(not sure if it compiles, partly inspired by #9981 but with adaptations) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23272 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-24Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have ↵kpfleming1-0/+5
autoconf and menuselect tools for Asterisk! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22267 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-03bug updates chan_h323 to some api changes it mogorman1-14/+13
also allows for it to compile once more patch taken from 6446 with minor mods, similar bugs 6521 and 6560 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17104 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-31define a global null_frame object so when queueing a null frame, you don'trussell1-3/+2
have to allocate one on the stack git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9001 f38db490-d61c-443f-a65b-d21fe96a405b
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