aboutsummaryrefslogtreecommitdiffstats
AgeCommit message (Collapse)AuthorFilesLines
2009-11-04Merged revisions 227739 via svnmerge from file1-417/+1922
https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227743 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 227372 via svnmerge from qwell2-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227373 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227368 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227361 via svnmerge from lmadsen1-12/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227362 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
2009-11-03Blocked revisions 227238 via svnmergedvossel0-0/+0
........ 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@227242 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 227162 via svnmerge from lmadsen1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227163 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
2009-11-02SIP channel name uniquenessdbrooks0-0/+0
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@226975 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Merged revisions 226890 via svnmerge from file1-4/+27
https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226891 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Merged revisions 226812 via svnmerge from tilghman1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226813 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-29Merged revisions 226532 via svnmerge from file2-2/+10
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
2009-10-28Merged revisions 226384 via svnmerge from lmadsen1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226387 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226378 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226381 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226305 via svnmerge from tilghman1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226306 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-27Merged revisions 226159 via svnmerge from tilghman1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226167 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-26detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabitzafrir2-8860/+8172
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 Merged revisions 226018 via svnmerge from http://svn.digium.com/svn/asterisk/trunk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226019 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-26Blocked revisions 225912 via svnmergejpeeler0-0/+0
........ 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@225916 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-26Backport audio handling loop fixes from trunk version of app_fax.kpfleming1-18/+23
This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (issue #16127) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225869 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-23Merged revisions 225582 via svnmerge from kpfleming1-3/+27
https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225583 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225485 via svnmerge from lmadsen2-10/+45
https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225486 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225360 via svnmerge from tilghman3-6/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225361 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 dvossel4-7/+47
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
2009-10-21Merged revisions 224932 via svnmerge from russell7-115/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224933 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Merged revisions 224856 via svnmerge from tilghman1-4/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224857 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Merged revisions 224774 via svnmerge from file1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224775 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224671 via svnmerge from kpfleming1-9/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224672 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224567 via svnmerge from file1-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224568 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224448 via svnmerge from tilghman1-2/+18
https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes. This corrects an issue reported on the -users list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224449 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
2009-10-15Merged revisions 224178 via svnmerge from jpeeler1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224179 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223832 via svnmerge from jpeeler1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223833 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223756 via svnmerge from dvossel1-19/+36
https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223759 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223652 via svnmerge from kpfleming2-3/+55
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
2009-10-11Merged revisions 223487 via svnmerge from russell1-5/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223488 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 223330 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223331 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223273 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223276 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223215 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223226 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 223136 via svnmerge from mnicholson1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223172 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
2009-10-09Merged revisions 223088 via svnmerge from dvossel1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223091 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222880 via svnmerge from russell4-68/+38
https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222881 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-22 02:49:26 +0000