mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
17403 Commits (3c9667ae121d23fe6efe01543bcdadeb4fc09eff)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
fa389ab17a |
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322) Reported by: amessina git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
550f7f1e65 |
Merged revisions 173967-173968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
|
647b68ec23 |
Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
b2484d7db9 |
Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
5da26420e9 |
Blocked revisions 173900 via svnmerge
........ r173900 | tilghman | 2009-02-06 09:43:32 -0600 (Fri, 06 Feb 2009) | 3 lines Backport OS X fix from trunk (AGAIN, closes issue #14360) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
0cbada4f96 |
Add a common implementation of a scheduler context with a dedicated thread.
This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
1d46bb0d0f |
Resolve a memory leak that would occur on an invalid channel given to Action: Status
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
69dff2f5f8 |
Update extensions.conf.sample to be correct.
In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
8f9c8f8b40 |
Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage
(closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
bf5358dd26 |
Blocked revisions 173770 via svnmerge
........ r173770 | mmichelson | 2009-02-05 17:19:16 -0600 (Thu, 05 Feb 2009) | 20 lines Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests With this fix, we only will perform an SRV lookup at the following times: * The first time we register with a remote registrar * If we send a REGISTER but do not receive a response * If the sendto() function returns an error While I wrote the patch that fixes this issue, a huge amount of credit is due to Brett Bryant, who wrote the initial patch on which I based this one. (closes issue #12312) Reported by: jrast Patches: 12312.patch uploaded by putnopvut (license 60) Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
4ac807845d |
Merged revisions 173696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
2e1d9f9a21 |
Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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4cc1606d27 |
Change the first field, or we don't get the necessary field separation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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5d3fc1ee3a |
Merged revisions 173592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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5f83636af4 |
Merged revisions 173559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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172777bd02 |
Fix some areas where the incorrect interface was passed to ast_device_state
I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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f931abc61a |
Add XML documentation for the applications and functions in res_jabber
(closes issue #14405) Reported by: snuffy Patches: xml_jabber.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
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b605bdff0c |
Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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39ec5d1576 |
Merged revisions 173211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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f2fd6528a2 |
When using a socket as a FILE *, the stdio functions will sometimes try to do
an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e04e940a71 |
Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173397 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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668f9f4e31 |
Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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bd244342e2 |
Fix a problem where file playback would cause fds to remain open forever
The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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f90021fdd0 |
Ensure that commas placed in the middle of extension character classes do not
interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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f8f032e98b |
Blocked revisions 173248 via svnmerge
........ r173248 | dvossel | 2009-02-03 17:35:55 -0600 (Tue, 03 Feb 2009) | 6 lines Fixes issue with IAX2 transfer not handing off calls. Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
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e70fc3d136 |
Broke up the large conditional blocks so it is easy to see if a function is compiled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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e179e613f7 |
1. Make OS X compile cleanly with app_stack.
2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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673d85387a |
Merged revisions 173070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
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34be09bf5c |
Merged revisions 173066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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c668cbfbfc |
Reverting commit number 173028 as there are some
potential issues git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173047 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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7db67f9ca7 |
Add a CLI command to log out a manager user
(closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Richard Mudgett
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8be58b59c1 |
Recorded merge of revisions 172962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) | 11 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172963 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Steve Murphy
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53d9b77898 |
This reverts the changes I made for 11583; will
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17 years ago |
Leif Madsen
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fdcc0a9a60 |
Update the res_ldap.conf file with a better working example.
(closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Steve Murphy
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c61e8a7865 |
This change allows the disconnect feature (as in "one-touch" in features.c)
to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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326587ebe3 |
Fix a spelling mistake.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
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a9ee30da54 |
Add a todo. I do need to really check what's going on with this kill-the-user business ;-)
Why do we suddenly have two flags to set peer type? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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81a3d40c08 |
Small formatting change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Olle Johansson
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9ea148b260 |
Add some well-needed improvements to the wishlist in the code, so that we can close
some bug reports. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
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2af8f59958 |
The CID lookup feature wasn't actually working properly with dialog-info+xml
supporting devices. The devices (snoms, specifically) need to receive a SIP URI instead of just an extension. This adds that functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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317e4e7313 |
Blank argument crashes Asterisk
(closes issue #14377) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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a2ddc0bb5e |
Don't increment the loop, now that incrementing is taken care of by the
decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Terry Wilson
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27862b0dce |
Blocked revisions 172639 via svnmerge
........ r172639 | twilson | 2009-01-30 18:15:09 -0600 (Fri, 30 Jan 2009) | 5 lines Rename new parkedcallparking option to parkedcallreparking Since this option actually already existed in 1.6.0+, use the same name so as not to confuse people when they upgrade ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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458dfe4748 |
Fix redefinition of flag in channel.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Terry Wilson
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a010aa5ade |
Remove incorrect line from sample config
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Terry Wilson
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8d782f96b8 |
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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402c61117a |
Parameter position reversed in documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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80f91f6ab7 |
Merged revisions 172438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Richard Mudgett
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3b08de23df |
Remove tabs from comment
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Richard Mudgett
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97b4e9cf2a |
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |