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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
2007-04-13A fix for chan_oss that resulted from the CDR changes; it helps to use the ↵murf1-1/+1
right info. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61644 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09This is a big improvement over the current CDR fixes. It may still need ↵murf1-5/+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
2007-04-08Off by one error, resulting in a crash (Issue 9500)tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60709 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-23Update channel drivers to use module referencing so that unloading them ↵file1-11/+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
2007-01-19Merge the changes from the /team/group/vldtmf_fixup branch.russell1-3/+4
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
2006-12-24Fix a typo in an error message that indicated that the MGCP channel type couldrussell1-1/+1
not be registered, instead of the correct type, OSS. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48948 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07These mods are to solve the problem in bug 7506. It's a lot of rework to ↵murf1-4/+6
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
2006-10-04update 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
2006-09-21Some more clean up in the load function for chan_oss (issue #8002 reported ↵file1-20/+20
by Mithraen with minor mods by moi) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43456 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-49/+380
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-08Formatting fixes for chan_oss (issue #7808 reported by Mithraen)file1-241/+211
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42389 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/+10
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):kpfleming1-15/+3
- 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 therussell1-2/+1
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
2006-07-24fix uninitialized variablerizzo1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38148 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵kpfleming1-0/+22
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
2006-07-05Merged revisions 36998 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r36998 | file | 2006-07-05 11:31:01 -0400 (Wed, 05 Jul 2006) | 2 lines Spell extension correctly in documentation for chan_oss dial (issue #7487 reported by flefoll) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36999 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-03Blocked revisions 36725 via svnmergerussell1-6/+3
........ 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 dynamicallyrussell1-12/+9
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 structurerussell1-9/+12
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 ↵kpfleming1-9/+7
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,russell1-9/+5
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
2006-05-31move the includes of abstract_jb.h to be with the rest of the asterisk includes.russell1-2/+2
These used to be wrapped in a #ifdef git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31078 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-31Add support for using a jitterbuffer for RTP on bridged calls. This includesrussell1-0/+48
a new implementation of a fixed size jitterbuffer, as well as support for the existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov) Thank you very much to Slav Klenov of Securax and all of the people involved in the testing of this feature for all of your hard work! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31052 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10ensure that control frames with payload can be sent to channel drivers via ↵kpfleming1-2/+2
->indicate() update iax2_indicate to pass control frame payload to the connected channel add an API call for sending an indication with payload, and use it for control frames with payload git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26417 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-09Make menuselect be able to show what the modules aremarkster1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25929 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-30use proper method of detecting OSS support on Linux and FreeBSDkpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23736 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-24Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have ↵kpfleming1-0/+6
autoconf and menuselect tools for Asterisk! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22267 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-9/+6
As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20003 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
2006-03-30don't use 8000, use DEFAULT_SAMPLE_RATErizzo1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16560 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28another batch of cli simplificationsrizzo1-13/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15857 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28fix documentationrizzo1-16/+43
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15514 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-10Add boost gain option to microphone, also minor bug fixes (#6388)mattf1-18/+85
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9449 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-08Merged revisions 9232 via svnmerge from mogorman1-0/+3
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r9232 | mogorman | 2006-02-08 16:12:34 -0600 (Wed, 08 Feb 2006) | 4 lines Make logger report error,warning,notice if logger.conf not found, also updated chan_oss to give correct error message if its config file is not found. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9239 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-01use string fields for some stuff in ast_channelkpfleming1-19/+16
const-ify some more APIs remove 'type' field from ast_channel, in favor of the one in the channel's tech structure allow string field module users to specify the 'chunk size' for pool allocations update chan_alsa to be compatible with recent const-ification patches git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9060 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-18constify arguments in more places where strings should not be modified ↵russell1-1/+1
(issue #6286) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8203 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-11convert some channels to use the memory allocation wrappers.russell1-10/+9
(This is being added to the janitor projects list.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7954 f38db490-d61c-443f-a65b-d21fe96a405b
2005-12-30update doxygen docs to specify authorsrussell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7682 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-29git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7221 ↵kpfleming1-0/+0
f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-21issue #5800russell1-0/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7158 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-12fix a lil' typorussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7090 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-06issue #5605russell1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6979 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-06Fix chan_oss buglets (bug #5618)markster1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6967 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-27Remove unnecessary checks before calls to ast_strlen_zero. Also, changerussell1-3/+3
some places where strlen is used instead of ast_strlen_zero git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6866 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-24Doxygen documentation update from oej (issue #5505)russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6847 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-16Make alsa/oss ignore VIDUPDATE control framesmarkster1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6799 f38db490-d61c-443f-a65b-d21fe96a405b
2005-09-27Fix "console" help (thanks to Andy Dvorak)markster1-2/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6670 f38db490-d61c-443f-a65b-d21fe96a405b
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