asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	c6f89f7ca3	Merged revisions 290674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290674 \| dvossel \| 2010-10-06 16:22:51 -0500 (Wed, 06 Oct 2010) \| 4 lines Fixes commented out code to use #if 0 instead. Thanks to rmudgett for catching this! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3a986a75c1	Merged revisions 290648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290648 \| dvossel \| 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) \| 12 lines Fixes gtalk outbound DTMF to work properly. Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered this after investigating a packet capture from the gmail client. Instead of performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue with negotiating RTP payload type 106 for the telephony-event and then sending DTMF as payload 101. This has been resolved by always negotiating 101 as the payload type like we do everywhere else. With this patch, incoming google voice calls forwarded to Asterisk via gtalk work. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ae6e8ecfd2	Merged revisions 290506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290506 \| dvossel \| 2010-10-05 17:23:00 -0500 (Tue, 05 Oct 2010) \| 2 lines Fixes uninitialized memory problem in 'iax2 set debug peer' option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	268ae2e8d5	Merged revisions 290479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290479 \| dvossel \| 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) \| 6 lines Fixes chan_gtalk to work with gmail client This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a8e290cd15	Merged revisions 290378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r290378 \| dvossel \| 2010-10-05 15:09:06 -0500 (Tue, 05 Oct 2010) \| 11 lines Resolves dnsmgr memory corruption in chan_iax2. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: afried, russell, dvossel Review: https://reviewboard.asterisk.org/r/965/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	c44527e185	Merged revisions 289840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289840 \| jpeeler \| 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) \| 29 lines Merged revisions 289798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 \| jpeeler \| 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) \| 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 \| jpeeler \| 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) \| 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bb485fc6f9	Merged revisions 289701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289701 \| jpeeler \| 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) \| 28 lines Merged revisions 289700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 \| jpeeler \| 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) \| 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 \| jpeeler \| 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) \| 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Stefan Schmidt	15cb4412f8	don't iterate through all dialogs to find and delete old subscribes On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	72fbcfd95d	Merged revisions 289554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289554 \| mnicholson \| 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) \| 11 lines Merged revisions 289553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 \| mnicholson \| 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) \| 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	8193e24e1a	Merged revisions 289549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289549 \| rmudgett \| 2010-09-30 14:28:36 -0500 (Thu, 30 Sep 2010) \| 17 lines Merged revision 289547 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r289547 \| rmudgett \| 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) \| 10 lines In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted. The same thing happens with DivertingLegInformation1 DivertedTo number. The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened PartyNumber field unconditionally. It now checks the presented number unscreened type to see if the PartyNumber was even present. JIRA ABE-2595 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	01eda62762	Merged revisions 289057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289057 \| rmudgett \| 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) \| 5 lines Avoid deadlock processing incoming AOC-E messages. Deadlock avoidance for the owner channel was not done when processing incoming AOC-E messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	8bbe682e45	Merged revisions 289054-289055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289054 \| rmudgett \| 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) \| 1 line Break up long ast_manager_event_multichan() event lines. ........ r289055 \| rmudgett \| 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) \| 1 line Revert stuff not ready for commit in -r289054. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c60da4ec9d	For an INVITE transaction, treat all 2XX responses the same as a 200. ABE-2305 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	9860ca7d16	Formatting fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	475cd60ab2	Merged revisions 288961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288961 \| tilghman \| 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) \| 5 lines Still build SIP, even if res_crypto cannot be built (use, not depend). (closes issue #18062) Reported by: a user on the mailing list ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9b8cdd8a9f	Merged revisions 288852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288852 \| dvossel \| 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) \| 5 lines Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	344bd58d56	Merged revisions 288821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288821 \| dvossel \| 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) \| 4 lines Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3. ABE-2293 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	4e473de5e2	Merged revisions 288748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288748 \| twilson \| 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) \| 19 lines Merged revisions 288747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288747 \| twilson \| 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) \| 12 lines Merged revisions 288746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288746 \| twilson \| 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) \| 5 lines Don't fail a masquerade if it is already being hung up This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	5ad9625cbf	Merged revisions 288507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288507 \| twilson \| 2010-09-22 16:18:27 -0700 (Wed, 22 Sep 2010) \| 22 lines Merged revisions 288500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288500 \| twilson \| 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) \| 15 lines Merged revisions 288499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 \| twilson \| 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) \| 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a2a1ec5336	Merged revisions 288418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288418 \| dvossel \| 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) \| 18 lines Merged revisions 288417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 \| dvossel \| 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) \| 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 \| dvossel \| 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) \| 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e6382a2dcb	Merged revisions 288345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288345 \| dvossel \| 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) \| 16 lines Merged revisions 288344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 \| dvossel \| 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) \| 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 \| dvossel \| 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) \| 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c5f5c24103	Merged revisions 288194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288194 \| rmudgett \| 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) \| 40 lines Merged revisions 288193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288193 \| rmudgett \| 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) \| 33 lines Merged revisions 288192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288192 \| rmudgett \| 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) \| 26 lines In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel. Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	949e81e6e5	Merged revisions 288159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288159 \| tilghman \| 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) \| 29 lines Merged revisions 288113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 \| tilghman \| 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) \| 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 \| tilghman \| 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) \| 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b287e93101	Merged revisions 288157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288157 \| pabelanger \| 2010-09-21 18:26:15 -0400 (Tue, 21 Sep 2010) \| 15 lines Merged revisions 288147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288147 \| pabelanger \| 2010-09-21 18:22:43 -0400 (Tue, 21 Sep 2010) \| 9 lines Setup timer before set_config(). (closes issue #18019) Reported by: Netview Patches: issue_0018019.patch uploaded by pabelanger (license 224) Tested by: Netview ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Stefan Schmidt	ee5af946e2	Instead of iterate through all dialogs, add two separte container for needdestroy and rtptimeout adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks. both container will be checked on every loop of do_monitor instead of iterate through all dialogs. (closes issue #17912) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	08aeb74d7a	Merged revisions 287929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287929 \| dvossel \| 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) \| 4 lines Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header. ABE-2258 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4a356afb7d	Merged revisions 287895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287895 \| russell \| 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) \| 10 lines Don't use ast_strdupa() from within the arguments to a function. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9b4cfb0d28	Merged revisions 287893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287893 \| tilghman \| 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) \| 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f92fd39b5c	Merged revisions 287683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287683 \| rmudgett \| 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) \| 9 lines The inalarm flag was not set in sig_analog struct if the port is initially in alarm. Fixed initial inalarm value for sig_analog ports. Along with -r261007, this gets the inalarm flag in sync with chan_dahdi for sig_analog ports. (closes issue #16983) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e2d002a144	Merged revisions 287645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287645 \| dvossel \| 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) \| 9 lines Fixes issue with registrations not working properly with pedantic=yes. (closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	27fbd5e156	Merged revisions 287643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287643 \| qwell \| 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) \| 15 lines Merged revisions 287642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287642 \| qwell \| 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) \| 8 lines Don't crash when parking a non-bridged call. (closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	7c77cebd4e	We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incoming SIP messages. Adding error based on RFC 3398 recommendations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c1af98603b	Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 \| rmudgett \| 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) \| 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 \| rmudgett \| 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) \| 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 \| jpeeler \| 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) \| 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b3fa5ec3be	Merged revisions 286904-286905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286904 \| rmudgett \| 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) \| 12 lines Unable to originate calls using E&M over T1. When originating a call from Unit Under Test to Reference Unit using E&M RBS signaling mode, I get the following warning message: "Ring/Off-hook in strange state 3 on channel 1". Fixed the sig_analog outgoing flag. It was never set when sig_analog was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ r286905 \| rmudgett \| 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) \| 1 line Simplify some code in sig_analog. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f9c7f53a1f	Merged revisions 286868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286868 \| mnicholson \| 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) \| 16 lines Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c994bfae3d	Merged revisions 286834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286834 \| dvossel \| 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) \| 2 lines Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2bb5307c8d	Merged revisions 286758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286758 \| mnicholson \| 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) \| 27 lines Merged revisions 286757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 \| mnicholson \| 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) \| 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 \| mnicholson \| 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) \| 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7b2c877fcb	Merged revisions 286457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286457 \| qwell \| 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) \| 12 lines Merged revisions 286456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 \| qwell \| 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) \| 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	a6480ff889	Formatting changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	d04046fbe7	Merged revisions 286189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286189 \| twilson \| 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) \| 30 lines Merged revisions 286115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 \| twilson \| 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) \| 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 \| twilson \| 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) \| 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b51f922a34	Merged revisions 286120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286120 \| pabelanger \| 2010-09-10 17:11:08 -0400 (Fri, 10 Sep 2010) \| 18 lines Merged revisions 286117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286117 \| pabelanger \| 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) \| 11 lines Merged revisions 286114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 \| pabelanger \| 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) \| 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1efb27a045	Merged revisions 286118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286118 \| rmudgett \| 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) \| 25 lines Merged revisions 286116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286116 \| rmudgett \| 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) \| 18 lines Merged revisions 286113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 \| rmudgett \| 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) \| 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	83bc091ac3	Merged revisions 285568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285568 \| dvossel \| 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) \| 16 lines Merged revisions 285567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285567 \| dvossel \| 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) \| 9 lines Merged revisions 285566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 \| dvossel \| 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) \| 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ede9032f92	Merged revisions 285564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285564 \| dvossel \| 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) \| 60 lines Merged revisions 285563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285563 \| dvossel \| 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) \| 54 lines Fixes interoperability problems with session timer behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ \|SITUATIONS \| 'session-timer' MODES \| \|___________\|________________________\| \| \| originate \| accept \| \|-----------\|------------\|-----------\| \|1. \| Active \| Inactive \| \|2. \| Active \| Active \| \|3. \| XXXXXXXX \| Active \| \|4. \| XXXXXXXX \| Inactive \| \|5. \| Active \| XXXXXXXX \| -------------------------------------- (closes issue #17005) Reported by: alexrecarey ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	dc7e1c6183	Merged revisions 285455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285455 \| qwell \| 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) \| 8 lines Don't automatically add domains for wildcard bindaddrs. (closes issue #17832) Reported by: oej Patches: 17832-wildcard.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	9b6fac435b	Merged revisions 285369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285369 \| qwell \| 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) \| 7 lines Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress. (closes issue #17831) Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6c5e3d5966	Merged revisions 285195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285195 \| rmudgett \| 2010-09-07 12:47:34 -0500 (Tue, 07 Sep 2010) \| 20 lines Merged revisions 285193 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 285192 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........ r285192 \| rmudgett \| 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) \| 8 lines COLP/CONP and chan_misdn missing update chan_misdn does not update the caller id of the channel if a new connected number or ECT-INFORM (w/ new peer number on call transfer) is received. JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	3b5727bf38	Merged revisions 285017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285017 \| twilson \| 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) \| 4 lines Call correct lock function as transferer is a sip_pvt not a channel Both functions are #defined to ao2_lock, but still... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1b2039e7db	Merged revisions 285006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 \| dvossel \| 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) \| 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6760 Commits (c6f89f7ca3eaa751e66da598331a4b60500d16dc)