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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
2010-09-02Merged revisions 284666 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284666 | tilghman | 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) | 9 lines Merged revisions 284665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines Fixing build. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284610 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284597 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after last commit ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-01Merged revisions 284561 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines During request to dialog matching, verify init_ruri is present before comparing. During request to dialog matching, we attempt a best effort routine for fork detection which requires several elements to be in place. The dialog's initial request uri is one of those elements. Since it is best effort, if the init_ruri is not present for some reason we can not proceed with that routine. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-01Merged revisions 284477 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-31Merged revisions 284415 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-31Add trustrpid and sendrpid global values to 'sip show settings'Leif Madsen
(closes issue #17860) Reported by: jtodd Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10) Tested by: lmadsen, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-27Merged revisions 284032 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-26Merged revisions 283692 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-25Merged revisions 283595 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-25Merged revisions 283559 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-25Merged revisions 283527 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24Ignore redial hard button when no previous number.Damien Wedhorn
(closes issue #17887) Reported by: salecha Patches: skinny.redial.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24Merged revisions 283493 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24Merged revisions 283457 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines Fix issue where TOS is no longer set on RTP packets. Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk. (closes issue #17890) Reported by: elguero Patches: qos_18.diff uploaded by elguero (license 37) Review: https://reviewboard.asterisk.org/r/868 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24Merged revisions 283382 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23Hack to allow easy debugging of skinny in trunk.Damien Wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23Add additional AST_CONTROL_ states to control2str.Damien Wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23Fixes display issues on 7910 and older phones.Damien Wedhorn
Also correct the callinfo provided in skinny_answer. (closes issue #17876) Reported by: salecha Patches: skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-20Merged revisions 283050 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines Merged revisions 283049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-20Merged revisions 282638 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-19Merged revisions 282895 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-19Merged revisions 282891 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines fixes sip peer memory leaks in the peer_by_ip table (issue #17798) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-19Merged revisions 282860 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-18Cleanup: consolidate offhook (new call).Damien Wedhorn
Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one. (closes issue #17812) Reported by: wedhorn Patches: cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-18Merged revisions 282671-282672 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line Use the correct operator when calculating the PRI span devstate. ........ r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line Use the correct type for aoce_delayhangup bit field. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-18Merged revisions 282639 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests. This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received. (issue #17486) Reported by: davidw Tested by: mnicholson (issue #12713) Reported by: davidw Review: https://reviewboard.asterisk.org/r/860/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-18Merged revisions 282608 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines Merged revisions 282607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-17Merged revisions 282577 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines Merged revisions 282576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines fixes no default transport for temp peer creation in chan_sip (closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-17Merged revisions 282545 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 Aug 2010) | 6 lines ACCEPT message should respond with the new FORMAT2 ie (closes issue #17804) Reported by: tpanton ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-15Support for GNU/kFreeBSDTzafrir Cohen
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING This patch gets Asterisk close to building on Debian kFreeBSD i386, mainly by adding an extra test for __GLIBC__ in one or two (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support (and support for chan_vpb) was not tested. Review: https://reviewboard.asterisk.org/r/858/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-14Merged revisions 282366 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines Fix our FRACKing issue with chan_iax2 a different way. Review: https://reviewboard.asterisk.org/r/861/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282334 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282302 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282269 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines res_stun_monitor for monitoring network changes behind a NAT device Review: https://reviewboard.asterisk.org/r/854 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282236 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines Merged revisions 282235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines only do magic pickup when notifycid is enabled A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-11Merged revisions 281874 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines handle all possible responses to REFER requests (closes issue #17486) Reported by: davidw Patches: Issue17486-counterbid.diff.txt uploaded by davidw (license 780) Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-11Merged revisions 281870 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1 only. * Also a couple minor tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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