asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	2bc3964ec9	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/1.6.0@227278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b6560b3586	Blocked revisions 227238 via svnmerge ........ 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/1.6.0@227242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	c1cc4d0833	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/1.6.0@227168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	5002373306	Merged revisions 227162 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2400c2cc61	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/1.6.0@227102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	3264cb328c	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/1.6.0@226976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	42d27c4390	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/1.6.0@226975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ae224e9a75	Merged revisions 226890 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	29a2e11d90	Merged revisions 226812 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	12b962ac19	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/1.6.0@226533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	d1c3865674	Merged revisions 226384 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	0c583527bc	Merged revisions 226378 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	69b4362386	Merged revisions 226305 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1c7cbc50a7	Merged revisions 226159 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tzafrir Cohen	4a76a51630	detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7c19321b0d	Blocked revisions 225912 via svnmerge ........ 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/1.6.0@225916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d84d683dbd	Backport audio handling loop fixes from trunk version of app_fax. 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d54d256d78	Merged revisions 225582 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	9e5c03795d	Merged revisions 225485 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b8b4e92f99	Merged revisions 225360 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	1a0047d161	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/1.6.0@225311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	31c282574b	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/1.6.0@225310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ffe8346471	Merged revisions 224932 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0f2eec5ab0	Merged revisions 224856 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2edc76e63c	Merged revisions 224774 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	43805797d7	Merged revisions 224671 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e23fe41857	Merged revisions 224567 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	db344967da	Merged revisions 224448 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	474544e477	fix typo, sorry git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	b597aca06a	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/1.6.0@224332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	e2c27b5cd0	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/1.6.0@224262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7473efd695	Merged revisions 224178 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e0dad69187	Merged revisions 223832 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e61e6c49f8	Merged revisions 223756 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	09fe2f94e6	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/1.6.0@223653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	aff8f71b32	Merged revisions 223487 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d50c91589f	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/1.6.0@223404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9a90216324	Merged revisions 223330 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	7f1d6e1b02	Merged revisions 223273 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	97136b0957	Merged revisions 223215 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	78405dbefe	Merged revisions 223206 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	83bc41dcb9	Merged revisions 223136 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	c56235b3c2	Merged revisions 223132 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a52d30e560	Merged revisions 223088 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f51bc4532e	Merged revisions 222880 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4afafcf938	Merged revisions 222873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 \| dvossel \| 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) \| 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	e3ada2755d	Merged revisions 222799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 \| rmudgett \| 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) \| 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 \| rmudgett \| 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) \| 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	171b954b9e	Merged revisions 222692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 \| rmudgett \| 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) \| 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 \| rmudgett \| 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) \| 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f3b41c917f	Properly initialize ast_devstate_aggregate so we don't crash sporadically. This looks like it was just missed during a merge. (closes issue #15841) Reported by: amorsen Patches: ast_devstate_aggregate_init-in-ast_extension_state2.patch uploaded by amorsen (license 676) Tested by: amorsen (closes issue #15852) Reported by: amorsen Tested by: amorsen, farisraouf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4de944c06c	Merged revisions 222543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 \| dvossel \| 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) \| 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 \| dvossel \| 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) \| 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

1 2 3 4 5 ...

17566 Commits (2bc3964ec9a5a985ec0a8cbbf391357082f379bc) All Branches Search

17566 Commits (2bc3964ec9a5a985ec0a8cbbf391357082f379bc)

All Branches