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2007-01-23Update channel drivers to use module referencing so that unloading them ↵file1-8/+1
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.russell1-2/+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
2007-01-05Merged revisions 49635 via svnmerge from kpfleming1-9/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49636 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-22Note to self: Run make before committing...qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48888 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-22Fix for issue 7774 - patch by alamantiaqwell1-4/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48870 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-10Discussion of these CLI changes resulted in more consistency (Bug 8236)tilghman1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47436 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-08Committed in behalf of 8190.murf1-5/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47344 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/+2
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" ↵tilghman1-1/+1
(continuation of issue 8236) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47053 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-02Reverse change of "show" to "list" and make several other commands more ↵tilghman1-35/+10
consistent with "category verb arguments" git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47051 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-09Fix a problem where phones that go "missing" never got unregistered.qwell1-48/+70
Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44764 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04update thread creation code a bitkpfleming1-2/+2
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
2006-09-28Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43933 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-26Add proper codec support to chan_skinny. Works with at least ulaw, alaw, ↵qwell1-15/+96
and g729a. This is technically a "new feature", but there are justifications for it. I found a bug with the recent rtp packetization changes, which caused the media setup to fail under certain circumstances, particularly when using allow=all, or having no allow= statements (globally or on the device). I could have either removed the rtp packetization features, or I could add proper codec support (which, without, I think most people would consider to be a bug anyways). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43650 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-22Allow chan_skinny.so to be unloaded properly.qwell1-18/+55
Remove reload support, since it doesn't actually...work. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43518 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-22First shot at unload_module in chan_skinny..qwell1-37/+25
More to come. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43469 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18allow for packetization on rtp channel drivers, need to addmogorman1-20/+24
option for setting our own packetization as apposed to just doing what is asked. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43243 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-28/+52
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-17remove a bit of redundant codeqwell1-31/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43123 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-17Skinny hold support.qwell1-4/+114
Original patch by wedhorn, with modifications by me. Issue #7588 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43111 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-17Get rid of all the useless (and confusing) typedefsqwell1-210/+210
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43099 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-02It was pointed out to me, that using two frames is silly.qwell1-13/+11
Thanks Josh. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41790 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-02fix some module loader breakageqwell1-11/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41789 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-02Fix VLDTMF issues.qwell1-7/+21
It still isn't variable length, but it does let you dial again. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41788 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31everything that loads a config that needs a config file to runmogorman1-1/+3
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 ↵file1-4/+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
2006-08-28Merge in RTP-level packet bridging. Packet comes in, packet goes out - ↵file1-14/+18
that's what RTP-level packet bridging is all about! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41235 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21Fix a potential integer signedness problem.qwell1-0/+7
Also fix some locking issues I found at the same time. Issue 7770, original patch by alamantia git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40757 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-15/+8
- 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
2006-08-20This is just step 1 in taking over the world.qwell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40663 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-20Converted device2str and control2str to use thread local storage.qwell1-4/+15
Thanks Russell. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40600 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-20Make the CIPC device string usefulqwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40563 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-20Fix minor display issues in "skinny show devices".qwell1-7/+75
Patch by wedhorn with minor modifications by me. Issue 7766 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40561 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-16move the calls to ast_jb_configure() to before the PBX thread is started on therussell1-4/+3
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
2006-08-05Merged revisions 38903-38904 via svnmerge from russell1-1/+7
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
2006-07-21Merge a new implementation of ast_inet_ntoa, our thread safe replacement forrussell1-11/+7
inet_ntoa, which uses thread specific data (aka thread local storage) instead of stack allocatted buffers to store the result. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38042 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵kpfleming1-5/+11
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
2006-07-18move variable declarations to the beginning of a block.rizzo1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37907 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-14Rename a couple variables, to be consistent with the rest of the functionsnorth1-108/+124
Abstract out some common code into a single function With the recent scheduler "issues", it pointed out a few things I might have been missing, so I added some rudimentary vrtp and rtcp stuff General cleanup... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37629 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-03Blocked revisions 36725 via svnmergerussell1-26/+17
........ 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
2006-07-01It might help if this had the right stimulus namenorth1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36562 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-23fix some little issues on big endian platforms to resolve compiler warningsrussell1-9/+13
- don't call the byte swapping macros on single byte numbers - don't do a ++ increment in the argument in the argument to the byte swapping macros. This gets expanded to incrementing the variable 4 times in a single operation, which results in undefined (and obviously undesired) behavior. :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35766 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-23revert my changes that converted the jb on the channel to be dynamicallyrussell1-16/+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/+16
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
2006-06-20fix an issue with dtmf not being sent sometimesnorth1-7/+15
slightly modify handling of digits git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35032 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-18No need to have debug enabled by default anymore..north1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34717 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-18Finally merge chan_skinny fixes into trunk.north1-1291/+2256
ready...set...GO! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34696 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-4/+4
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,russell1-8/+2
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
2006-06-01This is what I get for coding while tired.. sorry ;/north1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31385 f38db490-d61c-443f-a65b-d21fe96a405b
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