21735 Commits (b18d634377b132002ac9a65b14cd3e178c1d76b6)

 

All Branches   

Search

Author	SHA1	Message	Date
Richard Mudgett	b18d634377	Merged revisions 339504,339506 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines Add missing documentation of required AMI action Challenge AuthType header. (closes issue ASTERISK-18554) Reported by: Vlad Povorozniuc Patches: __20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI action Challenge. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	2e35748372	Merged revisions 339505 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ee29743c1f	Only change the capabilities on the gateway when ... the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4ced768c56	Merged revisions 339406 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines Make always create the MOH directory (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael Keuter ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	692ef8fdbf	Merged revisions 339352 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines Removes improper use of sound 'and' in German language mode from application saynumber Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk will only say the 'and' word between the single digit and double digit places. (closes issue ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	9c1362a6d1	Merged revisions 339297 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	e7801b07e9	Merged revisions 339244 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines fix forget declaration in previous change ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	cd49ac4da3	Merged revisions 339147 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	15f3a3bee1	Merged revisions 339144 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	3a6d2b3600	Merged revisions 339087 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines destroy memheap mutex properly before memheap deleted (fix memory leak occured after r304950 changes with DEBUG_THREAD compile option) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	050f405fbc	Merged revisions 339086 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e71f06475e	Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. ... This function prints a list of caps instead of a hex bitfield. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	36b780871a	Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	0469ffa481	properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	63c8b3336e	Documentation noting the extension of CHANNEL() for chan_ooh323 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ef74f3592e	Remove the channel function OOH323() and place its options into ... CHANNEL() channel drivers should not have there own dialplan functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	16cf441dcc	Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will ... turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	283f63c594	Remove T38 Gateway capability when detaching framehook. ... SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	af0161689f	Merged revisions 338800 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	681d28f83d	Merged revisions 338718 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line Adds documentation for QueueMemberStatus event generation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	67749ae7bb	Fix formatting of AMI header for SIP show peer. ... ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	c3cbe3478c	Merged revisions 338555 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines Test modules should depend on the TEST_FRAMEWORK flag ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6eb8315095	Merged revisions 338551 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line Test modules have a support level of core. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	dd73844f72	Merged revisions 338492 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	540b7a38a9	Merged revisions 338416 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5d5dc6650d	Merged revisions 338322 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	58ddd59548	Merged revisions 338235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	351d8edd83	Merged revisions 338227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e77f0fd5e	Merged revisions 338224 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	80a5b370ff	Merged revisions 338084 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0764556d4f	Merged revisions 337973 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e4b65a77b1	Merged revisions 337898 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines Spelling fix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	876cf3f78e	Merged revisions 337839 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	c62350dd70	Merged revisions 337774 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d95dcd14e3	Merged revisions 337720 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	cf4d5e585a	Revert previous commit ... New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	afe021b267	Forgot to svn add new files to r337595 ... Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	857e4fdb14	Generate Security events in chan_sip using new Security Events Framework ... Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	34ae017509	Merged revisions 337541 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c28595382c	Merged revisions 337486 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ee9db5269c	Revert commit r337261 ... This commit is for trunk not version 10 ----- Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. ----- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4272dcbb1a	Merged revisions 337430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	94d511c5fa	More silly spacing changes ...	14 years ago
Tilghman Lesher	e71a6090f2	........ ...	14 years ago
Tilghman Lesher	e964770729	........ ...	14 years ago
Gregory Nietsky	614479dd24	Whitespace fixup from SRTP patch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	edecb53861	Adds a timeout argument to app_originate ... the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	725dbbab22	Make ast_pbx_run() not default to s@default if extension is not found ... Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	42c967eea9	Change strictrtp option to default to yes in the RTP module ... Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	944cdaa94d	Merged revisions 337118 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago