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2010-10-02Merged revisions 289840 via svnmerge from Jeff Peeler
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
2010-10-01Merged revisions 289701 via svnmerge from Jeff Peeler
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
2010-10-01don't iterate through all dialogs to find and delete old subscribesStefan Schmidt
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
2010-09-30Merged revisions 289554 via svnmerge from Matthew Nicholson
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
2010-09-30Merged revisions 289549 via svnmerge from Richard Mudgett
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
2010-09-28Merged revisions 289057 via svnmerge from Richard Mudgett
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
2010-09-28Merged revisions 289054-289055 via svnmerge from Richard Mudgett
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
2010-09-27For an INVITE transaction, treat all 2XX responses the same as a 200.David Vossel
ABE-2305 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-27Formatting fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-27Merged revisions 288961 via svnmerge from Tilghman Lesher
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
2010-09-24Merged revisions 288852 via svnmerge from David Vossel
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
2010-09-24Merged revisions 288821 via svnmerge from David Vossel
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
2010-09-24Merged revisions 288748 via svnmerge from Terry Wilson
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
2010-09-22Merged revisions 288507 via svnmerge from Terry Wilson
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
2010-09-22Merged revisions 288418 via svnmerge from David Vossel
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
2010-09-22Merged revisions 288345 via svnmerge from David Vossel
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
2010-09-22Merged revisions 288194 via svnmerge from Richard Mudgett
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
2010-09-21Merged revisions 288159 via svnmerge from Tilghman Lesher
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
2010-09-21Merged revisions 288157 via svnmerge from Paul Belanger
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
2010-09-21Instead of iterate through all dialogs, add two separte container for ↵Stefan Schmidt
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
2010-09-21Merged revisions 287929 via svnmerge from David Vossel
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
2010-09-21Merged revisions 287895 via svnmerge from Russell Bryant
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
2010-09-21Merged revisions 287893 via svnmerge from Tilghman Lesher
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
2010-09-20Merged revisions 287683 via svnmerge from Richard Mudgett
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
2010-09-20Merged revisions 287645 via svnmerge from David Vossel
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
2010-09-20Merged revisions 287643 via svnmerge from Jason Parker
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
2010-09-16We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incomingOlle Johansson
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
2010-09-15Merged revisions 287017 via svnmerge from Richard Mudgett
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
2010-09-15Merged revisions 286931 via svnmerge from Jeff Peeler
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
2010-09-15Merged revisions 286904-286905 via svnmerge from Richard Mudgett
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
2010-09-15Merged revisions 286868 via svnmerge from Matthew Nicholson
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
2010-09-14Merged revisions 286834 via svnmerge from David Vossel
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
2010-09-14Merged revisions 286758 via svnmerge from Matthew Nicholson
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
2010-09-13Merged revisions 286457 via svnmerge from Jason Parker
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
2010-09-11Formatting changes.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-10Merged revisions 286189 via svnmerge from Terry Wilson
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
2010-09-10Merged revisions 286120 via svnmerge from Paul Belanger
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
2010-09-10Merged revisions 286118 via svnmerge from Richard Mudgett
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
2010-09-08Merged revisions 285568 via svnmerge from David Vossel
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
2010-09-08Merged revisions 285564 via svnmerge from David Vossel
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
2010-09-07Merged revisions 285455 via svnmerge from Jason Parker
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
2010-09-07Merged revisions 285369 via svnmerge from Jason Parker
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
2010-09-07Merged revisions 285195 via svnmerge from Richard Mudgett
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
2010-09-03Merged revisions 285017 via svnmerge from Terry Wilson
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
2010-09-03Merged revisions 285006 via svnmerge from David Vossel
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
2010-09-03Merged revisions 284967 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284967 | bbryant | 2010-09-03 14:19:53 -0400 (Fri, 03 Sep 2010) | 15 lines Merged revisions 284958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) | 8 lines This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response. (closes issue #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by alexkuklin (license 1115) Tested by: alexkuklin ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 284952 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) | 2 lines During OPTIONS authentication, the authpeer does not need to be returned for any reason. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 284950 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284779-284780 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) | 8 lines Made output libpri event names if pri debugging is enabled when sig_pri processes them. * Simplified CLI "pri debug xx span xx" command code and removed redundant debugging enabled messages. * Made CLI "pri debug xx span xx" command only close the debugging log file if it was opened. ........ r284780 | rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines Simplified pri_dchannel() poll timeout duration code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284705 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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