asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Nicholson	5b8a1e8bc5	Display the capabilities requested when requesting a fax session fails 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	2a7d090ba1	Merged revisions 297824 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8bd94df72a	Merged revisions 297819 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	461c3de2ed	Merged revisions 297713 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	00143b2778	Merged revisions 297605 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	cc870a3b31	Merged revisions 297534 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	f847ff2c38	Print a DEBUG message instead of a WARNING message when the selected fax 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	93454c0e30	Add support for reserving a fax session before answering the channel. Note: this change breaks ABI compatibility. FAX-217 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	ed77fa1dfe	Merged revisions 297405 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	165ec9b4e6	Merged revisions 297311 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3433890c9a	Merged revisions 297229 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	26e5a0d111	Changed some NOTICE and WARNING messages to DEBUG messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	add7816848	Merged revisions 297073 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@297075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5e42199e7b	Merged revisions 296991 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a41c5537d9	Merged revisions 296950 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	38b81d2772	Merged revisions 296869 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	82ee0bc14e	DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	9194596f7d	Merged revisions 296671 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	7017473c8c	Complete some error handling in transmit_publish() in chan_sip.c. 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b8249ee177	Merged revision 296575 from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5211e1c9d3	Merged revisions 296533 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4f92dec2c	Merged revisions 296466 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	dfbc5b89f9	Also don't build DEBUG_FD_LEAKS when STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	2bf0ccd7eb	Merged revisions 296351 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brad Watkins	81bb08714f	Fix XMPP PubSub-based distributed device state. 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brad Watkins	ee5d9d0835	Fix reloading of peer when a user is requested. 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f8153e4567	Merged revisions 296221 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5634ae11d5	Merged revisions 296166 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	515c5f489f	Merged revisions 296083 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	30a7e71c27	Merged revisions 296001 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@296002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	47cb712df7	Merged revisions 295907 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@295949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	f2901f8cca	Merged revisions 295868 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@295869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c08103f033	Merged revisions 295843 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.8@295866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a77913a22d	One way audio before answering call waiting call on analog port. * Analog call waiting Caller ID spills could get stuck resulting in one way audio until the waiting call is answered. This only happens on the second (and later) call waiting call if the active call is not the first call. * The CLI/AMI "dahdi show channel" command could report the wrong channel information. Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer in sync. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5153fbef97	Merged revisions 295710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295710 \| russell \| 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) \| 29 lines Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	1930461fa7	Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 \| twilson \| 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) \| 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 \| twilson \| 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) \| 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ddb80391f6	Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7fbafbd5df	Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed initial value of struct analog_pvt.use_callerid. It may get forced on depending upon other config options. * Call analog_dnd() instead of manual inlined code. * Removed unused struct analog_pvt.usedistinctiveringdetection. * Removed the struct analog_pvt.unknown_alarm flag. It was really the struct analog_pvt.inalarm flag. * Use ast_debug() instead of ast_log(LOG_DEBUG). * Rename several function's index variable to idx. * Some formatting tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b007bcc9b0	'sip notify clear-mwi' needs terminating CRLF. (closes issue #18275) Reported by: klaus3000 Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	f284b66dda	Merged revisions 295440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295440 \| pabelanger \| 2010-11-18 12:51:34 -0500 (Thu, 18 Nov 2010) \| 4 lines Fix compiler warnings when using openssl-dev 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	66e95a72d3	Add RedHat specific dependencies git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	14bf83e9d3	Uncomment settings under [global], to surpress warning when loading Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b63471b048	Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 \| rmudgett \| 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) \| 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 \| rmudgett \| 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) \| 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	73547b0961	Check for pdftotext and give a useful error if not found. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6a35147c4f	Remove intentional typo I had added when testing the check. oops. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	49750bb6e8	Check for wikiexport.py in PATH and give a useful error message if not found. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	583ca101ad	Merged revisions 295062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295062 \| tilghman \| 2010-11-15 12:24:02 -0600 (Mon, 15 Nov 2010) \| 9 lines Merged revisions 295026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295026 \| tilghman \| 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) \| 2 lines Create test verifying results of expression parser ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ab199924ac	Merged revisions 294988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 \| tilghman \| 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) \| 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	f1abd401b9	Merged revisions 294910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 \| jpeeler \| 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) \| 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	06ac20454e	Merged revisions 294904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 \| jpeeler \| 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) \| 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 \| jpeeler \| 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) \| 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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