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2011-03-16Merged revisions 310993 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-16Merged revisions 310902 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-15Merged revisions 310781 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) | 10 lines core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-14Merged revisions 310636 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-14Fixes null reference bug introduced by audio hook changes that affects ↵Jonathan Rose
various OS distributions. Thanks David. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-11Mix Monitor: Now with r and t options.Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-11Merged revisions 310287 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-10Merged revisions 310240 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines Add \r\n to remaining http headers passed to ast_http_send r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-07Merged revisions 309808 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-05Merged revisions 309678 via svnmerge from Tilghman Lesher
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
2011-03-02Add HangupRequest manager event, to specify when/where a channel gets hung up.Jason Parker
(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
2011-03-01Merged revisions 309204 via svnmerge from Jason Parker
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
2011-02-28Merged revisions 309035 via svnmerge from Tilghman Lesher
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
2011-02-24Merged revisions 308903 via svnmerge from Richard Mudgett
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
2011-02-24Merged revisions 308815 via svnmerge from Terry Wilson
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
2011-02-24Merged revisions 308723 via svnmerge from Matthew Nicholson
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
2011-02-23Fix compiler warning.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
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
2011-02-22Use ast_debug for console loggingAndrew Latham
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
2011-02-21Merged revisions 308416 via svnmerge from Matthew Nicholson
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
2011-02-21Add HTTP URI Debug logging and update noticeAndrew Latham
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
2011-02-21fix a memory leak in device stateTzafrir Cohen
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
2011-02-19Add CSS MIME TypeAndrew Latham
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
2011-02-15Merged revisions 307879 via svnmerge from Richard Mudgett
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
2011-02-10Merged revisions 307536 via svnmerge from Jason Parker
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
2011-02-10Fixes bug in chan_sip where nativeformats are not set correctly.David Vossel
The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Merged revisions 307273 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Allow parkedmusicclass to be settable for non-default parking lots.Jeff Peeler
(closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Merged revisions 307228 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09clarify warning when no loadable module supportTzafrir Cohen
Clarify warning message when LOADABLE_MODULES is disabled but we still try to load a module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Merged revisions 307142 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines Initialize tracking variable in structure properly. Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by me.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 307092 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines Fix issue with verbose messages not showing on remote console. This code was reworked recently, and since the logchannel list hadn't been created yet at this point, and it was a verbose message, it was being dropped on the floor. Now it'll continue on to where it should be handled. (closes issue #18580) Reported by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 307065 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines Add a couple of useful channel variables for the CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine the channel and context that will be called when the recall occurs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Merged revisions 306674 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Merged revisions 306575 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines Rearrange a bit of code in the generic CC recall operation. By waiting to call the callback macro after the CC_INTERFACES, extension, priority, and context have been set, this information can be accessed more easily within the callback macro. Reported by Philippe Lindheimer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Send manager event for blackfilter only if it DOES NOT match.Jeff Peeler
The logic got reversed, oops. Works properly now when multiple blackfilters are present. (closes issue #18283) Reported by: telecos82 Patches: ast_managereventfilter.patch uploaded by telecos82 (license 687) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 306124 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Modify alignment of 'core show codecs', since the ID is no longer a huge int.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Fixes output of "core show codecs" to display image types correctly.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 305923 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replacing doc/* with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305247 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines Add alternative name for config option. The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Asterisk HTTP response Content-typeAndrew Latham
Address content type for BSD and other platforms (closes issue #18456) Reported by: alexo Patches: asterisk18_http.patch uploaded by alexo (license 1175) Tested by: alexo git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 304950 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) | 18 lines Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used. This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-28Merged revisions 304638 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan 2011) | 11 lines Restore some conditionals that we lost in r277814. There are some cases where ast_append_ha() is called with a NULL instead of a valid int pointer. So if we get a NULL, don't try to dereference it. (closes issue #18162) Reported by: imcdona Patches: issue0018162.patch uploaded by pabelanger (license 224) Tested by: enegaard ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-27Merged revisions 304554 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or cc_agent_dialstring) are empty. Test if the value pointer is not NULL instead of not ast_strlen_zero(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Merged revisions 304339 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011) | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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