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Author	SHA1	Message	Date
Moises Silva	0f207dce6e	Merged revisions 309720 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	798212c828	Merged revisions 309678 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	01a453351d	Add setvar option to calendaring ... Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f6c2ebff45	Merged revisions 309585 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	3ef45303e1	Merged revisions 309542 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file. Also, prepend a newline to traceback output so that the main error message is on it's own line. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8861d64ecc	Merged revisions 309495 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is never used ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	b20fecdbbb	Add support for defining hints from pbx_lua ... (closes issue #16024) Reported by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c1ba13c1ea	Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk). ... In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	438ef2625a	Merged revisions 309448 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132. (closes issue #18671) Reported by: Igels Patches: pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) Tested by: Igels ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Ruggles	3cda82a379	Merged revisions 309403 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	a7bfa10472	Add HangupRequest manager event, to specify when/where a channel gets hung up. ... (closes issue #18226) Reported by: clegall_proformatique Patches: asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	070cb4ef87	Merged revisions 309256 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	dc616cfe2c	Merged revisions 309204 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines Fix consistency of CRLFs on HTTP headers that get sent out. (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	ffe9e4acfc	Merged revisions 309170 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Added XML documentation for CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML documentation for CHANNEL(reversecharge). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	72260849b7	Merged revisions 309126 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8e603ab4e1	Merged revisions 309084 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e5dc4c2d8e	Merged revisions 309035 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	008aa0e3b8	Merged revisions 308991 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled. (closes issue #18815) Reported by: irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot (license 52) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	b6e37118c9	Merged revisions 308945 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	642d6c306c	Merged revisions 308903 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines Invalid read in ast_channel_set_caller_event(). Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	f88ef55d74	Merged revisions 308815 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	0658c5ec2e	Merged revisions 308723 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	5deb544d06	Merged revisions 308679 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	79e041f856	Fix compiler warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	2b063d4dca	Merged revisions 308622 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff ... -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	736133f874	Use ast_debug for console logging ... Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	dad4fecc03	Merged revisions 308416 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	3a0af560c8	Add HTTP URI Debug logging and update notice ... enable reporting of the request URI / URL in debugging change funny debug note to a serious note. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	54802a099c	fix a memory leak in device state ... The callback handle_statechange (pbx.c) fails to release its data pointer, leaking memory in the process. Reported by: tzafrir Patches: 18735_pbx_free_callback.diff uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/1110/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Andrew Latham	83035fdf68	Add CSS MIME Type ... Modern browsers are checking for the MIME Type of pages and in some cases will not load a file if the type is wrong. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	659e748376	Merged revisions 308288 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines A few more (copies of) files to ignore in this directory. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	ebbb2cae64	Merged revisions 308242 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines added g729onlyA option for announce only AnnexA g.729 codec in h.323 capabilities. Option can be global or per user/peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b79adb645e	Add more verbage to CLI command 'pri show channels' usage. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	c87f9c827e	Merged revisions 308150 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines Fix FreeBSD builds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	dd5df75bef	Merged revisions 308098 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines ifdef __linux__ keepalive variables also ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	551dac2eda	Merged revisions 308010 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	7b44cd8ec7	include tcp keepalive socket calls only on linux, freebsd and others ... don't have these options on sockets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4a48600231	Add CLI "pri show channels" command. ... List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b1db966684	Merged revisions 307962 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line Don't crash when forcing caller id. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	64ed1ba3e9	Fixes compile error in chan_phone for big endian ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b2ef13cb60	Merged revisions 307879 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e38fa2d3cd	Merged revisions 307837 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5c49abfcf3	Merged revisions 307793 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once. (issue #18156) Reported by: asgaroth Patches: 20110214__issue18156.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c0e33b03c3	Making trunk compile again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7800a1c330	Merged revisions 307750 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	bd32cb4ba5	lc not found - it's warning, not error, ... change malloc to ast_calloc again git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	eaf73d6588	change malloc to ast_calloc calls to prevent crash of asterisk ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	96cbd4ffcd	Merged revisions 307536 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'm reverting here. (issue #18776) Reported by: alecdavis ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago