You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

Jason Parker 5e65377992 Multiple revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 ... ........ r377136 | rmudgett | 2012-12-03 14:33:08 -0600 (Mon, 03 Dec 2012) | 17 lines Cleanup core main on exit. * Cleanup time zones on exit. * Make exit clean/unclean report consistent for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377166 | rmudgett | 2012-12-03 16:53:58 -0600 (Mon, 03 Dec 2012) | 15 lines Cleanup ast_run_atexits() atexits list. * Convert atexits list to a mutex instead of a rd/wr lock. The lock is only write locked. * Move CLI verbose Asterisk ending message to where AMI message is output in really_quit() to avoid further surprises about using stuff already shutdown. (issue ASTERISK-20649) Reported by: Corey Farrell ........ Merged revisions 377165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377212 | rmudgett | 2012-12-04 16:31:02 -0600 (Tue, 04 Dec 2012) | 1 line confbridge: Update online XML documentation. ........ r377227 | rmudgett | 2012-12-04 18:49:53 -0600 (Tue, 04 Dec 2012) | 29 lines confbridge: Fix several small issues. * Made func_confbridge_helper() allow an empty value when setting options. You previously could not Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the dialplan. * Made func_confbridge_helper() handle its datastore better if multiple threads attempt to set the first CONFBRIDGE option value on the channel. * Made the func_confbridge_helper() only output one diagnostic message concerning the option. * Made the bridge video_mode able to repeatedly change in the config file and CONFBRIDGE dialplan function. The video_mode option values are an enum and not independent of each other. * Made handle_cli_confbridge_show_bridge_profile() better handle the video_mode option. * Simplified datastore handling code in conf_find_user_profile() and conf_find_bridge_profile(). * Made parse_bridge(), parse_user(), and parse_menu() use var->file instead of CONFBRIDGE_CONFIG because the var could have been from an include file. (closes issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter ........ r377241 | rmudgett | 2012-12-04 20:09:13 -0600 (Tue, 04 Dec 2012) | 4 lines * Fix registering core show codecs/codec CLI commands twice. * Fix registering atexit format_attr_shutdown() more than once. ........ r377258 | file | 2012-12-05 10:49:33 -0600 (Wed, 05 Dec 2012) | 19 lines Fix a SIP request memory leak with TLS connections. During the TLS re-work in chan_sip some TLS specific code was moved into a separate function. This function operates on a copy of the incoming SIP request. This copy was never deinitialized causing a memory leak for each request processed. This function is now given a SIP request structure which it can use to copy the incoming request into. This reduces the amount of memory allocations done since the internal allocated components are reused between packets and also ensures the SIP request structure is deinitialized when the TLS connection is torn down. (closes issue ASTERISK-20763) Reported by: deti ........ Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377261 | jrose | 2012-12-05 10:57:26 -0600 (Wed, 05 Dec 2012) | 15 lines res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session When srtp_create fails, the session may be dealloced or just not alloced. At the same time though, the session pointer might not be set to NULL in this process and attempting to srtp_dealloc it again will cause a segfault. This patch checks for failure of srtp_create and sets the session pointer to NULL if it fails. (closes issue ASTERISK-20499) Reported by: tootai Review: https://reviewboard.asterisk.org/r/2228/ ........ Merged revisions 377256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377354 | rmudgett | 2012-12-06 17:56:45 -0600 (Thu, 06 Dec 2012) | 24 lines confbridge: Fix some resource leaks on conference teardown. * Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t. * Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can destroy them unconditionally. * Made join_conference_bridge() abort if the new conference could not be added to the conferences container. * Made leave_conference() discard any post-join actions if join_conference_bridge() had to abort early. * Made the join_conference_bridge() diagnostic messages better describe what happened. * Renamed leave_conference_bridge() to leave_conference() and made it only take a conference user pointer. The conference pointer was redundant. * Made conf_bridge_profile_copy() use struct copy instead of memcpy(). * No need to lock the conference in start_conf_record_thread() since all of the callers already have it locked. ........ r377382 | kmoore | 2012-12-07 15:58:21 -0600 (Fri, 07 Dec 2012) | 17 lines codec_dahdi: Fix output of "transcoder show" CLI command. In r306010 "Asterisk media architecture conversion - no more format bitfields", the logic for incrementing encoders and decoders when opening transcoder channels was changed without making the corresponding change when decrementing encoder / decoder channels. The result being that when a channel was destroyed, codec_dahdi couldn't properly tell if it was an encoder or decoder, and the default case is to assume it was a decoder. This could result in negative numbers for decoders in use like in: VOIP6*CLI> transcoder show 2/-2 encoders/decoders of 92 channels are in use. (closes issue ASTERISK-19921) Patch-by: Shaun Ruffell ........ r377399 | rmudgett | 2012-12-07 17:42:03 -0600 (Fri, 07 Dec 2012) | 5 lines MALLOC_DEBUG: Only wait if we want atexit allocation dumps. ........ Merged revisions 377398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377432 | rmudgett | 2012-12-07 18:29:23 -0600 (Fri, 07 Dec 2012) | 14 lines Fix order of SIP allow/disallow in MySQL contrib script. Using the contrib sippeers.sql script to create the sippeers MySQL table would result in being unable to place calls if you set the disallow value to all. (closes issue ASTERISK-20756) Reported by: Andre Luis Patches: sippeers.patch patch uploaded by Andre Luis ........ Merged revisions 377431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377504 | tilghman | 2012-12-09 19:24:41 -0600 (Sun, 09 Dec 2012) | 5 lines Remove some dead code and additionally handle a case that wasn't handled. ........ Merged revisions 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377510 | tilghman | 2012-12-09 19:39:58 -0600 (Sun, 09 Dec 2012) | 16 lines Improve documentation by making all of the colors used readable, no matter what the background color is. Dark blue on a black background is unreadable, as is yellow on a light background. This patch turns on the bright attribute for colors when on a dark background and turns *off* the bright attribute when the -W command line option is used (indicating a _light_ background). This ensures that text is readable in both cases. Patch by: tilghman Review: https://reviewboard.asterisk.org/r/2224 ........ Merged revisions 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377558 | igorg | 2012-12-09 23:04:36 -0600 (Sun, 09 Dec 2012) | 8 lines Fix crash on transfer initiated from insreeen menu on Unistim phones. Removed CDR-related code that moved to do_masquarade before. (closes issue ASTERISK-20417) Reported by: Rudolf Migalin ........ Merged revisions 377557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377592 | igorg | 2012-12-10 00:41:47 -0600 (Mon, 10 Dec 2012) | 9 lines Fix codec mismatch Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. (issue ASTERISK-20183) ........ Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377624 | kmoore | 2012-12-10 08:40:26 -0600 (Mon, 10 Dec 2012) | 14 lines Handle Session-Expires less than local Min-SE in 200 OK Ensure that a call is immediately torn down if a Session-Expires value received in a 200 OK is less than the local Min-SE. This also prevents Asterisk from allowing calls with Session-Expires below the RFC4028-mandated minimum (90s). (closes issue ASTERISK-20653) Review: https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore ........ Merged revisions 377623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377656 | kmoore | 2012-12-10 10:53:16 -0600 (Mon, 10 Dec 2012) | 14 lines Ensure ReceiveFax provides a CED tone via T.38 When using res_fax_digium, the T.38 CED tone was not being provided properly which would cause some incoming faxes to fail. This was not an issue with res_fax_spandsp since it does not strictly honor the send_ced flag and sends the CED tone whenever receiving a T.38 fax. (closes issue FAX-343) Reported-by: Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377705 | rmudgett | 2012-12-10 18:32:40 -0600 (Mon, 10 Dec 2012) | 14 lines Cleanup dnsmgr on exit. * Cleanup dnsmgr thread and CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377709 | rmudgett | 2012-12-10 19:00:05 -0600 (Mon, 10 Dec 2012) | 15 lines Cleanup event on exit. * Cleanup CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to event.c missed in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377741 | rmudgett | 2012-12-10 20:11:29 -0600 (Mon, 10 Dec 2012) | 19 lines Cleanup indications on exit. * Made ast_unregister_indication_country() unlink the found tone zone before selecting a new default_tone_zone to make it impossible to select the tone zone being unregistered again. * Ringcadence is no longer parsed twice in store_config_tone_zone(). * Cleanup CLI commands and destroy default_tone_zone on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r377772 | rmudgett | 2012-12-10 20:42:34 -0600 (Mon, 10 Dec 2012) | 13 lines Cleanup logger on exit. * Cleanup CLI commands, destroy verbosers and logchannels lists on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@378659 65c4cc65-6c06-0410-ace0-fbb531ad65f3		13 years ago
..
g722	Shuttle some bits around to address some gain issues with G.722.	17 years ago
gsm	Merged revisions 375273 via svnmerge from	13 years ago
ilbc	Merged revisions 373631,373633,373645 via svnmerge from	14 years ago
lpc10	Multiple revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083	14 years ago
speex	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff	15 years ago
Makefile	Merged revisions 373631,373633,373645 via svnmerge from	14 years ago
codec_a_mu.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_adpcm.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_alaw.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_dahdi.c	Multiple revisions 377136,377166,377212,377227,377241,377258,377261,377354,377382,377399,377432,377504,377510,377558,377592,377624,377656,377705,377709,377741,377772	13 years ago
codec_g722.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_g726.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_gsm.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_ilbc.c	Include iLBC source code for distribution with Asterisk	14 years ago
codec_lpc10.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_resample.c	The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly.	15 years ago
codec_speex.c	Merged revisions 328209 via svnmerge from	15 years ago
codec_ulaw.c	Merged revisions 328209 via svnmerge from	15 years ago
ex_adpcm.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_alaw.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_g722.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_g726.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_gsm.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_ilbc.h	Fix compile error in codec ilbc translator.	15 years ago
ex_lpc10.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_speex.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
ex_ulaw.h	Asterisk media architecture conversion - no more format bitfields	15 years ago
log2comp.h	Merged revisions 109648 via svnmerge from	18 years ago