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2007-03-08fix a compiler warning, and overwriting 'res' valuekpfleming1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58352 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08fix two cases where HTTP session file descriptors would not be closedkpfleming1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58351 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08If we receive ZT_EVENT_REMOVED, destroy the specified channel. russell3-21/+34
(issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58320 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07(This bug was reported to me by Kinsey Moore)russell1-9/+17
Merged revisions 58242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines Fix a problem where the Asterisk channel name could be that of the wrong IAX2 user for a call. This is because the first step of choosing this name is to look for an IAX2 peer that happens to have the same IP/port number that this call is coming from and assuming that is it. However, this is not always correct. So, I have made it change this name after authentication happens since at that point, we have an exact match. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58243 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07Ensure we have (or should have) at least one matching codec before ↵file2-2/+13
attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58240 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07Blocked revisions 58167 via svnmergerussell0-0/+0
........ r58167 | russell | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a misplaced block of code in the 1.2 version of the patch to fix issue #8977 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58168 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07Merged revisions 58164 via svnmerge from russell1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines If the channels acquired using the manager Redirect action are not up, then don't attempt to do anything with them. It could lead to weird behavior, including crashes. (issue #8977) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58165 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Merged revisions 58115 via svnmerge from murf1-1/+17
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58121 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Clarify the documentation of the dialout and sendvoicemail options.russell1-5/+6
(issue #9000, caio1982 and serge-v) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58119 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Merged revisions 58052 via svnmerge from oej1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines Change error message to proper message ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58053 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Return an error of transmit_response is called without a session.russell1-0/+6
(issue #9002) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58023 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Since chan_iax2 does not support reception of DTMF with duration ensure that ↵file1-0/+1
it is set to 0 on the frame. (issue #8521 reported by gdhgdh) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57914 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Don't create a listen channel and record the conference unless the option is ↵file1-17/+16
turned on. (issue #9204 reported by francesco_r) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57872 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Merged revisions 57869 via svnmerge from file1-8/+8
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57870 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Merged revisions 57825 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57826 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)file1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57798 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)file1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57770 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03Updated the regression testsmurf4-35/+41
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57707 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03Merged revisions 57648 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines Memory leak of a list, if call recording was abandoned ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57649 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03submitted patch for Georgian language, issue 9010, submitted by Alexander ↵dhubbard1-0/+389
Shaduri git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57620 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03add missing configuration template. Thanks to Lacy Moore on asterisk-users ↵russell1-0/+10
for pointing this out\! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57591 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02Update the check that is used to determine whether zaptel transcoder supportrussell2-6/+6
is present. The interface has changed. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57556 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02Merged revisions 57475 via svnmerge from file1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57477 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02Merged revisions 57458 via svnmerge from murf1-9/+20
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57473 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02I almost had comma escapes right, but 9184 points out the problem-- the ↵murf2-0/+42
escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57426 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-02Return the correct digit that interrupted the stream. This fixes exiting therussell1-0/+1
Background application when using the m option. (issue #9176, mjagdis) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57396 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merge changes from svn/asterisk/team/russell/sla_updatesrussell4-41/+171
* Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57364 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merged revisions 57317 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57318 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-011. Corrected a typo for www.etsi.org. Thank Patrick.transnexus1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57263 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28minor tweaks to the sla docsrussell2-6/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge more changes from svn/asterisk/team/russell/sla_updatesrussell2-53/+160
* Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57203 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Minor formatting changerussell1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57146 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge changes from svn/asterisk/team/russell/sla_updatesrussell2-9/+30
* Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57144 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merged revisions 57118 via svnmerge from murf1-2/+13
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57139 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merged revisions 57092 via svnmerge from file1-1/+10
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57093 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge current set of changes from svn/asterisk/team/russell/sla_updatesrussell2-151/+467
* Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57089 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Picky compiler...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57055 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Better handle timeouts when the individual speaks after everything has been ↵file1-3/+4
played but before the timeout ends. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57053 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28I was surprised that I had not yet downgraded missing goto targets and macro ↵murf1-10/+38
call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57049 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-27Fix voicemail email attachments. I missed the conversion of one of the linerussell1-2/+2
endings and there was an extra one where it should not have been. (issue #9128) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56975 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Picky, picky... show deprecation warning in application help, too (reported ↵tilghman2-2/+6
via list) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56922 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Restore the behavior of Asterisk 1.2 where if a device was not specified inrussell1-2/+2
alsa.conf, then we just use the system default, instead of creating our own default of hw:0,0. (issue #9139) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56888 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Merged revisions 56850 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 lines Obey the clearglobalvars option in extensions reload (or dialplan reload depending on your version). (issue #9146 reported by ramonpeek) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56856 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Fix a crash in my last change to iax2_indicate(). (issue #9150)russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56847 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Update app_record documentation to use new CLI command, core show file ↵file1-1/+1
formats. (issue #9151 reported by junky) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56839 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Use ast_strlen_zero to see if the language and/or context argument is not ↵file1-2/+2
present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56805 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Do more complete locking of the chan_iax2_pvt struct in the indicate callback.russell1-6/+17
(Problem brought up by Ben Smithurst on the asterisk-dev list) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56785 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Allow both of the show version files and core show file versions CLI ↵file1-4/+81
commands to work. (issue #9135 reported by mvanbaak) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56783 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Move a comment to be in the correct struct.russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@56740 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-03 16:34:48 +0000