5958 Commits (2ee78bf6e63634bbe7e5b2319f6334aa587d019e)

Author	SHA1	Message	Date
Tilghman Lesher	5264340224	One duplicate setting here (dead code). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@237967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	32cdfa1923	Merged revisions 237319 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@237321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	47c27e1953	Merged revisions 237136 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@237138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a8df43f94f	Merged revisions 236982 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@236984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f3c57275b8	Merged revisions 236802 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@236803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1cbc8dcb57	Merged revisions 236063 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@236065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0a39e69815	reverses minor sip registration regression ... reverses the changes caused by issue #15539. The issue reported was expected behavior. (issue #15539) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@235135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	cb5cce0650	Merged revisions 234526 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r234526 | oej | 2009-12-14 11:46:20 +0100 (Mån, 14 Dec 2009) | 16 lines Merged revisions 234492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines Stop sending 183's after call hangup. There where still cases where the 183 keep-alive mechanism would not stop sending 183's even though the Asterisk server had sent a final reply to the invite. EDVX-28 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@234533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9eab56995d	Merged revisions 234129 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r234129 | tilghman | 2009-12-10 10:24:26 -0600 (Thu, 10 Dec 2009) | 16 lines Merged revisions 234095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines When we receive no response at all to our INVITE, allow the channel to be destroyed. (closes issue #15627) Reported by: falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14) 20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14) Tested by: falves11 Review: https://reviewboard.asterisk.org/r/446/ (closes issue #15716) Reported by: dant (closes issue #16270) Reported by: corruptor (closes issue #15356) Reported by: falves11 (issue #16382) Reported by: lftsy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@234132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0ec3fc29e7	Merged revisions 233472 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@233474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2a47210650	Merged revisions 233394 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@233395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	ffe164d3f5	Merged revisions 232345 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@232347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d6ba1b4f47	Merged revisions 232230 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario causing code to execute during reload that should not. (issue AST-263) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@232231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	06f5b30343	Merged revisions 232091 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@232093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	47deec6289	Merged revisions 231692 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@231694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	25fe750a8f	Merged revisions 231602 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@231604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d4fecf8429	Merged revisions 230881 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	add84870a2	Merged revisions 230877 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines Merged revisions 230839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line Correct fix for issue #16268... the reporter's original patch was very close to correct. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	8c4c2e2753	Merged revisions 230773 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines Merged revisions 230772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines Ensure that SDP parsing does not ignore the last line of the SDP. (closes issue #16268) Reported by: sgimeno ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	331c8e1975	Merged revisions 230726 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines fixes iax2 show cache locking error, thanks alecdavis! (closes issue #16094) Reported by: alecdavis Patches: bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	466344f266	Merged revisions 230247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600 (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines Correct mistaken option name in error message. The configuration option for allowing hosts to make non-token-based calls is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5c370f146f	Merged revisions 230145 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | 15 lines Merged revisions 230144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 lines Respect the maddr parameter in the Via header. (closes issue #14446) Reported by: frawd Patches: via_maddr.patch uploaded by frawd (license 610) Tested by: frawd ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	09c7fb67c5	Merged revisions 230039 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 lines Fix a crash caused by two threads thinking they should both free the chan_local private structure when only one should. (closes issue #15314) Reported by: sroberts Patches: Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780) Tested by: davidw, lottc ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2c8c6a49d4	Merged revisions 229912 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix T.38 negotiation regression introduced with the SDP parser changes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	2e7ace8c28	Merged revisions 229750 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 | qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix mute toggling on OSS channels. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a362e93d35	Merged revisions 229168 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	e9cea50233	Reverted revision 202008. ... (closes issue #16175) Reported by: paul-tg git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	42048030e0	Merged revisions 229015 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) | 8 lines Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	bac2f04b1a	fixes segfault when transferring a queue caller ... In sip_hangup we attempted to lock p->owner after we set it to NULL. Thanks to fhackenberger for reporting the issue and submitting a patch. (closes issue 0015848) Reported by: fhackenberger Patches: digium_bug_0015848 uploaded by fhackenberger (license 592) Tested by: fhackenberger, lmadsen, TomS, shin-shoryuken, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a40c49579c	Merged revisions 228548 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	919783597e	Merged revisions 227238 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	fc208f59ad	Merged revisions 228145 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	382a3a7ea8	Merged revisions 228080 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	0d49e1196b	Modify the SDP parsing code to parse session and media level items separately. ... With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	cdc5621ed2	Merged revisions 227712 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9028b47fe2	Merged revisions 227275 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	9366e1f3f7	Merged revisions 227167 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	8a3dba806f	Merged revisions 227091 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	e4ab593e8a	SIP channel name uniqueness ... SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	bf7f97a3ff	Merged revisions 226532 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2e6c200b27	Merged revisions 225912 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8dd2ce2ace	Merged revisions 225650 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 | dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines Fixes an iterator memory leak and uninitialized memory ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4672e2805b	Merged revisions 225445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	c64aca3b91	Merged revisions 225307 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	219e2238a9	Merged revisions 225033 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	9d34d37a4b	fix typo, sorry ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7593e10437	Merged revisions 224331 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6c68619844	Merged revisions 224261 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1b54dbccc7	Merged revisions 223652 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	9f1bf0f9bd	Fix interpretation of PRIREDIRECTIONREASON set by chan_sip. ... This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago