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2011-10-03Merged revisions 339089 via svnmerge from Alexandr Anikin
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon, 03 Oct 2011) | 10 lines 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/trunk@339091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 339088 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339088 | twilson | 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) | 17 lines 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/trunk@339090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 339045 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct 2011) | 4 lines 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/trunk@339046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 339043 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct 2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 339011 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct 2011) | 2 lines 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/trunk@339021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 338997 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) | 1 line Documentation noting the extension of CHANNEL() for chan_ooh323 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 338995 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines 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/trunk@338996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-03Merged revisions 338950 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) | 14 lines 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/trunk@338951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-02Merged revisions 338904 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) | 8 lines 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/trunk@338905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-01Update "configure" based on r338139.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Merged revisions 338801 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) | 19 lines 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/trunk@338802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Formatting changes onlyOlle Johansson
--Denna och nedanstående rader kommer inte med i loggmeddelandet-- M channels/chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Merged revisions 338719 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338719 | jrose | 2011-09-30 13:55:27 -0500 (Fri, 30 Sep 2011) | 9 lines 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/trunk@338720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Fix formatting of AMI header for SIP show peer.Richard Mudgett
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 ........ Merged revisions 338664 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Preserve DTMF length in main/features.cOlle Johansson
Review: https://reviewboard.asterisk.org/r/1463/ A small part of much larger work with DTMF duration in Asterisk, funded by IPvision AS in Denmark. Thanks to irroot for the review! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Merged revisions 338556 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400 (Thu, 29 Sep 2011) | 9 lines 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/trunk@338557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Merged revisions 338552 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338552 | qwell | 2011-09-29 15:54:55 -0500 (Thu, 29 Sep 2011) | 9 lines 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/trunk@338553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Blocked revisions 338493 via svnmergeLeif Madsen
................ r338493 | lmadsen | 2011-09-29 13:32:28 -0500 (Thu, 29 Sep 2011) | 14 lines 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/trunk@338494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Merged revisions 338417 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines 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/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Add CLI command "cdr show pgsql status" based on "cdr mysql status"Olle Johansson
Review: https://reviewboard.asterisk.org/r/923/ Thanks all for the code reviews and feedback. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Just formatting.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338323 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines 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/trunk@338324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338253 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) | 14 lines 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/trunk@338284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338228 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines 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/trunk@338229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338225 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines 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/trunk@338226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Update CHANGES to reflect autopausebusy not being in Asterisk 10Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Add autopausebusy and autopauseunavail queue optionsTerry Wilson
Make it possible to autopause on a busy or unavailable response from a device. (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches: autopausebusy.txt by twilson Review: https://reviewboard.asterisk.org/r/1399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Updated for checking OSP Toolkit version 4.0.0.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Updated for OSP Toolkit 4.0.0.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-27Merged revisions 338085 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400 (Tue, 27 Sep 2011) | 9 lines 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/trunk@338086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-27Whitespace (red blobs) fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-26Merged revisions 337974 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines 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/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-23Merged revisions 337902 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337902 | irroot | 2011-09-23 21:18:14 +0200 (Fri, 23 Sep 2011) | 10 lines 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/trunk@337910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-23Merged revisions 337840 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337840 | irroot | 2011-09-23 10:39:22 +0200 (Fri, 23 Sep 2011) | 17 lines 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/trunk@337855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-23Merged revisions 337775 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337775 | russell | 2011-09-22 19:45:35 -0500 (Thu, 22 Sep 2011) | 18 lines 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/trunk@337776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337721 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines 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/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Blocked revisions 337433 via svnmergeGregory Nietsky
........ r337433 | irroot | 2011-09-22 08:42:42 +0200 (Thu, 22 Sep 2011) | 12 lines 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/trunk@337681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Blocked revisions 337640 via svnmergePaul Belanger
........ r337640 | pabelanger | 2011-09-22 14:43:35 -0400 (Thu, 22 Sep 2011) | 5 lines 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/trunk@337641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337595,337597 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines 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/ ........ r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines 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/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337542 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337542 | irroot | 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) | 14 lines 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/trunk@337543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337487 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines 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/trunk@337488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337431 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines 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/trunk@337432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21More silly spacing changesTilghman Lesher
..... Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged revisions 337380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21................Tilghman Lesher
........ Dumb little spacing fix. ........ Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ Merged revisions 337345 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21................Tilghman Lesher
........ Escape commas in keys and values, when keys and values are enumerated by commas. Review: https://reviewboard.asterisk.org/r/1433 ........ Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ Merged revisions 337342 from https://origsvn.digium.com/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21Merged revisions 337263 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | 1 line Whitespace fixup from SRTP patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21Merged revisions 337261 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) | 10 lines 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/trunk@337262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21Merged revisions 337219 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines 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/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21Merged revisions 337178 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines 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/trunk@337179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 337120 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines 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/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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