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2007-02-16Adding Realtime Text support (T.140) to AsteriskOlle Johansson
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-31Merged revisions 53046 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines Merged revisions 53045 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@53047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19Merged revisions 51311 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-01- Implement error handling in ast_append_haOlle Johansson
- Use this in chan_sip - Document ha functions in acl.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06Constify a bunch of usage strings for CLI commands.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10Merged revisions 47436 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006) | 2 lines Discussion of these CLI changes resulted in more consistency (Bug 8236) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-07A fair number of changes for the sake of bug 7506Steve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02Merged revisions 47051 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25Merged revisions 46200 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-04Merged revisions 44378 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines update thread creation code a bit 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: https://origsvn.digium.com/svn/asterisk/trunk@44379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-03bug #8076 check option_debug before printing to debug channel.Matt O'Gorman
patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-28Merged revisions 43933 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-21Merged revisions 43454 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2 lines Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-21Lots more removal of deprecated thingsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-18merge qwell's CLI verbification workKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31everything that loads a config that needs a config file to runMatt O'Gorman
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: https://origsvn.digium.com/svn/asterisk/trunk@41633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-28Merge in RTP-level packet bridging. Packet comes in, packet goes out - ↵Joshua Colp
that's what RTP-level packet bridging is all about! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-21merge new_loader_completion branch, including (at least):Kevin P. Fleming
- 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: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-16Merged revisions 40057 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r40057 | kpfleming | 2006-08-16 13:57:44 -0500 (Wed, 16 Aug 2006) | 2 lines don't allow AUEP responses to overflow the stack during a string copy (reported by Mu Security) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-16move the calls to ast_jb_configure() to before the PBX thread is started on theRussell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@39964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-05Merged revisions 38903-38904 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@38905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-21Merge a new implementation of ast_inet_ntoa, our thread safe replacement forRussell Bryant
inet_ntoa, which uses thread specific data (aka thread local storage) instead of stack allocatted buffers to store the result. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵Kevin P. Fleming
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13allow users of RTP to use G726-32 AAL2 packing even when RFC3551 packing has ↵Kevin P. Fleming
been requested (Sipura/Grandstream ATAs and others will need this) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-03Blocked revisions 36725 via svnmergeRussell Bryant
........ 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: https://origsvn.digium.com/svn/asterisk/trunk@36726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-23revert my changes that converted the jb on the channel to be dynamicallyRussell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@35746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-22- dynamically allocate the ast_jb structure that is on the channel structureRussell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵Kevin P. Fleming
again :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01- add the ability to configure forced jitterbuffers on h323, jingle,Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31update the rest of the channel drivers that use RTP so that their channelRussell Bryant
tech structures indicate that they create jitter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-10ensure that control frames with payload can be sent to channel drivers via ↵Kevin P. Fleming
->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: https://origsvn.digium.com/svn/asterisk/trunk@26417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-21more stncpy/ast_copy_string replacement.Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-14This rather large commit changes the way modules are loaded. Luigi Rizzo
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: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-08since the module API is changing, it's a good time to const-ify the ↵Kevin P. Fleming
description() and key() return values git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-05Bug 6873 - Finish moving from the non-threadsafe (and poor randomness) ↵Tilghman Lesher
rand() to threadsafe ast_random() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-11Merged revisions 9609 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r9609 | russell | 2006-02-11 14:23:20 -0500 (Sat, 11 Feb 2006) | 2 lines fix memory leak from not destroying the scheduler context on module unload ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-10Merged revisions 9404 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r9404 | kpfleming | 2006-02-10 14:38:59 -0600 (Fri, 10 Feb 2006) | 2 lines don't create monitor threads in detached mode, when we need to be able to pthread_join() them later if the module is unloaded (solve crash-on-unload problem for these channel modules) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-01use string fields for some stuff in ast_channelKevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@9060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-31define a global null_frame object so when queueing a null frame, you don'tRussell Bryant
have to allocate one on the stack git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-21Bug 5515 - Devicestate and API documentation updateTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-30update doxygen docs to specify authorsRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-20Major RTP fixes for using inbound SDP on outbound connection, get rid of Mark Spencer
old local rtp stuff... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-03Bug 5858 - Make the chanvars.c functions return a 'const char *'Tilghman Lesher
This should prevent us from unintentionally changing variable values when they're returned from pbx_builtin_getvar_helper. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29remove extraneous svn:executable propertiesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-06issue #5605Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-10-27Remove unnecessary checks before calls to ast_strlen_zero. Also, changeRussell Bryant
some places where strlen is used instead of ast_strlen_zero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-10-24Doxygen documentation update from oej (issue #5505)Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-09-14update MANY more files with proper copyright/license info (thanks Ian!)Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-09-09allow users of RTP to know when the peer endpoint is (apparently) behind a NATKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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