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2010-04-27Add missing file (pointed out by TheDavidFactor on #asterisk-dev) referenced ↵tilghman1-0/+42
by revision 239231. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259189 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Merged revisions 259104 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines Let compilation succeed warning-free when DONT_OPTIMIZE is turned off. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259105 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Merged revisions 259018 via svnmerge from mmichelson1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines Prevent Newchannel manager events for dummy channels. No Newchannel manager event will be fired for channels that are allocated to not match a registered technology type. Thus bogus channels allocated solely for variable substitution or CDR operations do not result in a Newchannel event. (closes issue #16957) Reported by: atis Review: https://reviewboard.asterisk.org/r/601 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Line 24 missed in compatibility fix in revision 233577diruggles1-1/+1
added a "fun:" prefix line 24 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258974 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Small error in the T.140 RTP port verbose log.lmadsen1-2/+2
(closes issue #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258934 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2.mnicholson3-290/+624
The fax session initilization code for T.38 faxes has been rewritten. T.38 session initialization was removed from generic_fax_exec, and split into two different code paths for receive and send. Also the 'z' option (to send a T.38 reinvite if we do not receive one) was added to sendfax. In the output of 'fax show sessions', the 'Type' column has been renamed to 'Tech' and replaced with a new 'Tech' column that will report 'G.711' or 'T.38'. Control of ECM defaults has been added to res_fax A 'fax show settings' CLI command has been added. Support of the new AST_T38_REQUEST_PARMS control method request to handle channels that have already received a T.38 reinvite before the FAX application is start has been added. Support for the 'fax show settings' command has been added to res_fax_spandsp and handling of the ECM flag has been slightly altered. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258896 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-25additional checking related to issue 17186may1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258855 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-25Don't pass zero length callerid to ooh323 stackmay1-1/+1
Don't pass zero callerid string to ooh323 stack because it can't encode this properly and can't generate setup message. (closes issue #17186) Reported by: vmikhelson Patches: zero_callerid_num.patch uploaded by may213 (license 454) Tested by: may213 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258838 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-25Merged revisions 258775 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines When StopMonitor is called, ensure that it will not be restarted by a channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258776 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Add another random function that does nothing to make the utils/ dir happy.qwell1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258685 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Fix previous commit.mnicholson1-18/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258675 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Make utils/ stuff *actually* compile this time.qwell2-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258674 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Let utils/ dir compile when DEBUG_THREADS is not enabled.qwell2-6/+59
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258673 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Merged revisions 193391,258670 via svnmerge from mnicholson3-7/+26
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines Set the proper disposition on originated calls. (closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson ........ r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines Fix broken CDR behavior. This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call(). (closes issue #16797) Reported by: VarnishedOtter Tested by: mnicholson ........ (closes issue #16222) Reported by: telles Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Add ast_event subscription unit test and fix some ast_event API bugs.russell2-96/+463
This patch introduces another test in test_event.c that exercises most of the subscription related ast_event API calls. I made some minor additions to the existing event allocation test to increase API coverage by the test code. Finally, I made a list in a comment of API calls not yet touched by the test module as a to-do list for future test development. During the development of this test code, I discovered a number of bugs in the event API. 1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple of different places. The API allows a subscription to all event types, but with IE parameters, just as if it was a subscription to a specific event type. However, the parameters were being ignored. This affected ast_event_check_subscriber() and event distribution to subscribers. 2) Some of the logic in ast_event_check_subscriber() for checking subscriptions against query parameters was wrong. Review: https://reviewboard.asterisk.org/r/617/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258632 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Pass interactive = 0 and fix a compile error.eliel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258595 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Remove ABI differences that occured when compiling with DEBUG_THREADS.qwell8-1630/+1515
"Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a loaded module was not (or vice versa). This also immensely simplifies the lock code, since there are no longer 2 separate versions of them. Review: https://reviewboard.asterisk.org/r/508/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Asterisk data retrieval API.eliel13-46/+4540
This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258517 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Add MEETMEBOOKID from r256019.russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258515 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Merged revisions 258432 via svnmerge from jpeeler1-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines Fix looping forever when no input received in certain voicemail menu scenarios. Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258433 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Missed this when reverting the bad version change in asterisk.tex.lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258387 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Fix change in asterisk.tex that got merged in after testing.lmadsen1-1/+1
(issue #17220) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258383 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Add ability to generate ASCII documentation from the TeX files.lmadsen14-10702/+10118
These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258351 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Add small documentation update to func_callcompletion.c.mmichelson1-1/+4
This directs users to documents which can help explain the concepts and configuration options settable with the function. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21IAXpeers output now matches SIPpeers format for manager (AMI).lmadsen3-14/+32
(closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21fixes issue with double "sip:" in header fielddvossel1-1/+1
This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258305 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Fix the \brief description in the res_calendar_*.c files.lmadsen3-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258265 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21fix whitespace issuejmls1-10/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258256 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added NEW ACTIONS entry for new MixMonitorMute AMI command.jmls2-1/+13
Added State and Direction variables for new MixMonitorMute AMI command. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258228 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added CHANGES entry for new MixMonitorMute AMI command.jmls1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added MixMonitorMute manager commandjmls6-14/+166
Added a new manager command to mute/unmute MixMonitor audio on a channel. Added a new feature to audiohooks so that you can mute either read / write (or both) types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. (closes issue #16740) Reported by: jmls Review: https://reviewboard.asterisk.org/r/487/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Add 'soft hangup' alias per Steve Johnson on asterisk-users.lmadsen1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258149 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Add example dialplan for dialing ISN numbers (http://www.freenum.org).lmadsen1-0/+44
Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258147 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Add missing 'useragent' field to sip-friends.sql file.lmadsen1-0/+1
(closes issue #17171) Reported by: thehar Patches: sip-friends.patch uploaded by thehar (license 831) Tested by: pabelanger, thehar git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258106 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Merged revisions 258029 via svnmerge from jpeeler1-20/+39
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258065 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Update supported file extensions in doxygen.lmadsen1-1/+1
Updated the doxygen \arg line after looking at the file for some other Asterisk documentation and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257988 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Change log message to match severity.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257949 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Don't consider a missing indications.conf to be a critical error.qwell1-1/+1
There were many changes in revision 176627 which would avoid the error that a missing config would have caused. Other than this, there are no other config files (including asterisk.conf, surprisingly) that are required. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257947 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Bad merge fixtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257883 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Blocked revisions 257856 via svnmergejpeeler0-0/+0
........ r257856 | jpeeler | 2010-04-19 14:09:46 -0500 (Mon, 19 Apr 2010) | 1 line make app_voicemail compile with IMAP_STORAGE ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257857 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Commit compromise I suggested on review 608.mmichelson1-0/+14
This allows for multiple SRV queries to be done from the dialplan for the same service on a single call while still allowing one to bypass the call to SRVQUERY if they so please. Taking action since no comments had been left for a while. This can easily be reverted if needed. External tests still pass. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257851 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Fix incomplete CDR merge from r195881twilson1-6/+10
Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257810 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-18Removing unused configuration parameterstilghman1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257768 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Merged revisions 257686 via svnmerge from dhubbard1-5/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257713 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Make sure to fail a monitor if we receive a negative response for a CC ↵mmichelson1-0/+10
SUBSCRIBE. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257646 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Enable PRI SERVICE message support in chan_dahdi for the 'national' switchtypedhubbard1-4/+7
Revision 1072 of libpri added SERVICE message support for the 'national' switchtype. The attached patch enables the use of 'pri service' CLI commands on dahdi channels that are configured for the 'national' switchtype. (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch uploaded by dhubbard (license 733) Tested by: elguero, dhubbard Review: https://reviewboard.asterisk.org/r/612/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257642 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257544 via svnmerge from tilghman3-40/+137
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257560 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257467 via svnmerge from tilghman1-0/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257493 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257426 via svnmerge from lmadsen1-27/+105
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257427 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257342 via svnmerge from lmadsen1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257343 f38db490-d61c-443f-a65b-d21fe96a405b
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