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2011-01-17AST-2011-001lmadsen4-855/+7
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2.1@302150 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17Create 1.8.2.1 from 1.8.2lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2.1@302106 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Importing release summary for 1.8.2 release.v1.8.2lmadsen2-0/+854
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2@301499 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Update ChangeLog, .version file, and remove summary files.lmadsen4-819/+38
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2@301495 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12------------------------------------------------------------------------lmadsen4-7725/+33082
r301221 | pabelanger | 2011-01-09 15:40:35 -0600 (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 [^] ........ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue 0018589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ [^] ........ ------------------------------------------------------------------------ git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2@301494 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Create Asterisk 1.8.2 from 1.8.2-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2@301456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Use autotagged externalsv1.8.2-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2-rc1@298191 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing release summary for 1.8.2-rc1 release.lmadsen2-0/+818
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2-rc1@298190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing files for 1.8.2-rc1 release.lmadsen3-0/+26972
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2-rc1@298189 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Creating tag for the release of asterisk-1.8.2-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.2-rc1@298188 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-11Correction to work with gatekeeper which don't send GK IDmay1-31/+57
Don't use GK ID if it's not presented in GK replies Extract GK ID not only in GK confirm but in GK register confirm also (issue #18401) Reported by: MrHanMan Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested by: may213, MrHanMan git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@298099 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-10Prevent a memcpy overlap in GENERIC_FAX_EXEC_SET_VARSmnicholson1-3/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@298054 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-10Merged revisions 298050 via svnmerge from tilghman4-2/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) | 11 lines Portability issue on OpenSolaris. Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@298051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Merged revisions 297960 via svnmerge from twilson1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297965 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Fixes issue with outbound google voice calls not working.dvossel1-5/+40
Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (closes issue #18412) Reported by: nevermind_quack Patches: fix uploaded by dvossel (license 671) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Don't crash after Set(CDR(userfield)=...) in ast_bridge_calltwilson1-3/+7
Instead of setting peer->cdr = NULL, set it to not post. (closes issue #18415) Reported by: macbrody Patches: patch-18415 uploaded by jsolares (license 1167) Tested by: jsolares, twilson git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297952 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-08Merged revisions 297908 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010) | 4 lines Use inheritance to get correct results for SIPFROMDOMAIN. (from an internal Digium discussion) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297909 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-08Display the capabilities requested when requesting a fax session fails ↵mnicholson1-7/+86
instead of displaying a hex value. Tweak the way fax stats are calculated so that all fax attempts and faliures are logged. Also make ensure faxes are either counted as completed or falied and never both. FAX-210 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297905 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297824 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297825 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297819 via svnmerge from tilghman4-21/+107
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600 (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) | 4 lines Use non-deprecated APIs for CoreAudio Review: https://reviewboard.asterisk.org/r/1040/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297821 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297713 via svnmerge from tilghman1-2/+8
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297733 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Merged revisions 297605 via svnmerge from jpeeler1-6/+30
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297607 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-03Merged revisions 297534 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri, 03 Dec 2010) | 3 lines The CLI command should not contain <placeholder>s, these are for descriptions. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297535 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-03Print a DEBUG message instead of a WARNING message when the selected fax ↵mnicholson1-17/+19
tech does not support reserving sessions. Answer the channel before quering it for t.38 support. This is necessary for the query to work properly over local channels. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297495 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Add support for reserving a fax session before answering the channel.mnicholson2-49/+174
Note: this change breaks ABI compatibility. FAX-217 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297486 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297405 via svnmerge from pabelanger1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500 (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec 2010) | 7 lines Resolve compile error under FreeBSD We now set _ASTCFLAGS+=-march=i686 for i386 processors, still allowing ASTCFLAGS to override the setting. Review: https://reviewboard.asterisk.org/r/1043/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297406 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297311 via svnmerge from twilson1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297311 | twilson | 2010-12-02 12:07:39 -0600 (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines Initialize offset for adaptive jitter buffer When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297312 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297229 via svnmerge from russell1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297245 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Changed some NOTICE and WARNING messages to DEBUG messages.mnicholson1-13/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297157 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 297073 via svnmerge from jpeeler1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@297075 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 296991 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600 (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines Clarify documentation on how we store codec preference lists. (closes issue #18397) Reported by: birgita ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296992 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 296950 via svnmerge from tilghman1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30 Nov 2010) | 2 lines Missed initializations caused startup errors on Mac OS X (and possibly others, too). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 296869 via svnmerge from jpeeler1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296870 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-30DOC: Conference number can be omitted; if omitted, all users in a meetme are ↵tilghman1-1/+1
listed. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296787 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revisions 296671 via svnmerge from pabelanger1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500 (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) | 5 lines Make sure nothing else is needed before destroying the scheduler. (closes issue #18398) Reported by: pabelanger ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296673 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Complete some error handling in transmit_publish() in chan_sip.c.russell1-0/+2
This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revision 296575 fromrmudgett2-2/+27
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY redirecting number and notification code, SETUP redirecting number) is also sent in PTMP/TE mode. It should only apply in PTMP/NT mode. The call setup proceeds but the network (Deutsche Telekom) reacts with ugly ISDN STATUS messages. Also don't send the redirecting number ie when PTP is also sending the DivertingLegInformation2 facility. The redirecting number ie is redundant and the network (Deutsche Telekom) complains about it. Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) JIRA ABE-2651 JIRA SWP-2537 .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296582 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revisions 296533 via svnmerge from tilghman4-2/+36
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines I love standards. There are so many to choose from. Except when there isn't one. Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296534 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-27Merged revisions 296466 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296467 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-27Also don't build DEBUG_FD_LEAKS when STANDALONE2 is defined.tilghman1-2/+2
(closes issue #18385) Reported by: cmaj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296429 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Merged revisions 296351 via svnmerge from oej1-39/+48
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre, 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines Fix bugs in saying numbers using the Swedish language syntax (closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296391 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Fix XMPP PubSub-based distributed device state.marquis1-43/+40
Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (closes issue #18272) Reported by: klaus3000 Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, Marquis Review: https://reviewboard.asterisk.org/r/1030/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296354 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Fix reloading of peer when a user is requested.marquis1-5/+13
Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot. (closes issue #18342) Reported by: nivek Patches: issue0018342p1.patch uploaded by nivek (license 636) Tested by: nivek Review: https://reviewboard.asterisk.org/r/1029/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296352 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296221 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296230 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296166 via svnmerge from rmudgett4-214/+349
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296167 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296083 via svnmerge from russell1-7/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296084 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296001 via svnmerge from russell2-4/+13
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@296002 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-23Merged revisions 295907 via svnmerge from oej1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis, 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines Fix support of saynumber(1,n) in the Swedish language (closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@295949 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295868 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov 2010) | 2 lines Change some documentation to suggest dahdi_monitor instead of ztmonitor. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@295869 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295843 via svnmerge from rmudgett5-59/+134
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@295866 f38db490-d61c-443f-a65b-d21fe96a405b
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