aboutsummaryrefslogtreecommitdiffstats
path: root/channels
AgeCommit message (Collapse)AuthorFilesLines
2010-02-02Fixes T38 crash with invalid FaxMaxDatagram fielddvossel1-36/+64
AST-2010-001 git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.22@244386 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 209400 via svnmerge from tilghman5-28/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238494 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238412 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238441 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238405 via svnmerge from dvossel1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238408 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Something clearly went wrong with a merge somewhere, because these are all ↵tilghman1-65/+0
duplicates (and therefore dead code). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@237966 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04Merged revisions 237319 via svnmerge from tilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@237320 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-02Merged revisions 237136 via svnmerge from oej1-0/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@237137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-30Merged revisions 236982 via svnmerge from tilghman1-1/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@236983 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-29Merged revisions 236802 via svnmerge from tilghman1-4/+20
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@236805 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-22Merged revisions 236063 via svnmerge from dvossel1-3/+9
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@236066 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15reverses minor sip registration regressiondvossel1-7/+14
reverses the changes caused by issue #15539. The issue reported was expected behavior. (issue #15539) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@235136 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-14Merged revisions 234526 via svnmerge from oej1-0/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@234528 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Merged revisions 234129 via svnmerge from tilghman1-6/+10
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 ........ 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@234131 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Merged revisions 233472 via svnmerge from dvossel1-1/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@233475 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Merged revisions 233394 via svnmerge from mnicholson1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@233397 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232345 via svnmerge from file1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@232346 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232091 via svnmerge from jpeeler1-1/+5
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@232092 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231692 via svnmerge from kpfleming1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@231693 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231602 via svnmerge from file1-3/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@231603 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Merged revisions 230881 via svnmerge from file1-36/+62
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230882 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Merged revisions 230877 via svnmerge from kpfleming1-10/+15
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230878 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Merged revisions 230773 via svnmerge from kpfleming1-4/+5
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230782 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-20Merged revisions 230726 via svnmerge from dvossel1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230729 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-15Merged revisions 230247 via svnmerge from kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230248 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Merged revisions 230145 via svnmerge from file1-2/+13
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230146 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Merged revisions 230039 via svnmerge from file1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@230040 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Merged revisions 229912 via svnmerge from file1-43/+50
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@229913 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10Merged revisions 229168 via svnmerge from dvossel1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@229234 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10Reverted revision 202006.mnicholson1-14/+2
(closes issue #16175) Reported by: paul-tg git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@229100 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Merged revisions 228548 via svnmerge from file1-0/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@228549 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Fix a logic flaw I introduced when I was testing stuff out.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@228479 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Fix a crash caused by freeing a dialog directly instead of using dialog_unref.file1-18/+3
(closes issue #16097) Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej (license 841) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@228415 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228145 via svnmerge from dbrooks1-0/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@228146 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228080 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@228081 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Modify the SDP parsing code to parse session and media level items separately.mnicholson1-466/+611
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227763 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Merged revisions 227712 via svnmerge from file1-2/+0
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227717 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227275 via svnmerge fromrmudgett1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227278 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227167 via svnmerge from file1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227168 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227091 via svnmerge from oej1-2/+13
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227102 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02SIP channel name uniquenessdbrooks1-1/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@226976 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-29Merged revisions 226532 via svnmerge from file1-2/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@226533 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225307 via svnmerge from dvossel1-10/+77
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@225311 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225033 via svnmerge from dvossel2-7/+29
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@225310 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17fix typo, sorryjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224337 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17Merged revisions 224331 via svnmerge from jpeeler1-2/+9
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@224332 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-16Merged revisions 224261 via svnmerge from rmudgett1-4/+12
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@224262 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223652 via svnmerge from kpfleming1-2/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@223653 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Fix interpretation of PRIREDIRECTIONREASON set by chan_sip.jpeeler2-2/+4
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@223404 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223206 via svnmerge from dvossel1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223210 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223132 via svnmerge from dvossel1-12/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223135 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-20 11:02:10 +0000