mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
22224 Commits (ebe2c33b7239b2881007efa3efd8d8a859f0b653)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
279873e8eb |
Add admin toggle mute all and participant count menu options to app_confbridge
This patch adds two new menu features to app_confbridge, admin_toggle_menu_ participants and participant_count. The admin action will globally mute / unmute all non-admin participants on a converence, while the participant count simply exposes the existing participant count function to the conference bridge menu. This also adds configuration options to change the sound played when the conference is globally muted / unmuted, as well as the necessary config hooks to place these functions in the DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin Reeves Tested by: Matt Jordan Patches: app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281) Review: https://reviewboard.asterisk.org/r/1518/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
4a125f45a0 |
Remove dead code since pri_grab() can never fail.
Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
2d67b1b378 |
Guarantee messages go into the right folders with multiple recipients
Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345488 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345489 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
a86037d959 |
Make FastAGI HANGUP show up in AGI debug output.
* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.
(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345432 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
bd4f81b51f |
Fix typo in sig_pri using wrong structure name.
It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.
(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
sig_pri.c.diff (License #5755) patch uploaded by zvision
........
Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345371 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
9e726d9cb4 |
Make queue log indicate if ADDMEMBER is paused for AMI and realtime.
* Add parameter to queue log ADDMEMBER to indicate if the member is
paused.
(closes issue ASTERISK-18645)
Reported by: garlew
Patches:
paused.diff (License #5337) patch uploaded by garlew
Tested by: rmudgett, garlew
Review: https://reviewboard.asterisk.org/r/1469/
........
Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345290 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
113612b9d6 |
Restore SIP DTMF overlap dialing method.
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
1cef6cf8cd |
Fix Progress spelling error in main/pbx.c.
(closes issue ASTERISK-18857)
Reported by: David M
Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
........
Merged revisions 345219 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345220 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
59d6db63bd |
Don't read past end of input when calling write()
int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when new_frames == 1. Otherwise we start reading into adjacent variables declared on the stack. The read end discards what is read, so the values don't matter but it's not a good idea to read past where we want even though new_frames is almost always 1 and should never be large. This patch is basically taken out of kpfleming's eventfd branch, as he mentioned that he remembered fixing it there when I talked to him about this issue. Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345164 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6ef49c3214 |
Update reqresp_parser parse_uri doxygen comments.
The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345161 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ec237d2e4a |
Moves voicemail setup password entry to the end of the setup process.
This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345117 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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818ac23b92 |
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345064 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f436a6f27c |
Increased max number of destinations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3845299e46 |
mISDN Round Robin break when no channel is available
Prevent channels been parsed repetitively. ........ Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344966 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bd486fcf41 |
Don't forget to rescan MOH files for cached realtime classes
Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ ........ Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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735e48f92f |
Use __alignof__ instead of sizeof for stringfield length storage.
Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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60f51c002a |
Video format was treated as audio when removed from the file playback scheduler
This patch fixes the format type check in ast_closestream and filestream_destructor. Previously a comparison operator was used, but since audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats that have a value greater than the video formats), a bitwise AND operation is used instead. Duplicated code was also moved to filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo Bedrij Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1580/ ........ Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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863d7189b9 |
Remove unneeded if(params) checks in reqresp_parser.
Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344839 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bac9ff62ef |
Fix bad quoting of multiline mxml opaque_data that caused invalid xml.
The opaque_data was added and enclosed in single quotes, assuming it would be only a single line. The rest of the lines were appended after the closing quote. (closes issue ASTERISK-18852) Reported by: peep_ on IRC Review: https://reviewboard.asterisk.org/r/1577 ........ Merged revisions 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344836 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a4365a8ae2 |
Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax transmission, fax teardown fails. Make sure capability is only touched if RTP is active. This regression was introduced in R344385. ........ Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344770 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e48cecc848 |
Check sip.conf maxforwards parameter for range 1 <= x <= 255.
JIRA AST-710 ........ Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344716 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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39beaff425 |
Make CLI "core show channel" not hold the channel lock during console output.
Holding the channel lock while the CLI "core show channel" command is executing can slow down the system. It could block the system if the console output is halted or paused. * Made capture the CLI "core show channel" output into a buffer to be output after the channel is unlocked. * Removed use of C++ keyword as a variable name. out renamed to obuf. * Checked allocation of obuf for failure so will not crash. (closes issue ASTERISK-18571) Reported by: Pavel Troller Tested by: rmudgett ........ Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344662 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8d994bed55 |
Fix a segmentation fault when using an extension with CID matching and no CID.
Attempting to call an extension which used Caller ID matching with a channel that has an empty caller id string would result in a segmentation fault. (closes issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged revisions 344608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344609 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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751488b84c |
Fix app_macro.c MODULEINFO section termination.
(closes issue ASTERISK-18848) Reported by: Tony Mountifield ........ Merged revisions 344557 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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46089f6b51 |
Fix potential deadlock calling ast_call() with channel locks held.
Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344540 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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464b337b3c |
Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.
It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344537 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0a2a79c94b |
Merged revisions 344493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344494 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dc05ce5e4f |
Fix another incorrect case with meetme's PIN logic and add documentation
This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344440 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c225800646 |
Fix several bugs with SDP parsing and well-formedness of responses
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3d44965e70 |
only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses
Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344334 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6d481420ce |
Fix deadlock during dialplan reload.
Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
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Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 344271 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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04f04ff39d |
Don't treat a host:port string as a domain
The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fa57bbe17c |
Add a unit test for ast_sockaddr_split_hostport
Review: https://reviewboard.asterisk.org/r/1575/ ........ Merged revisions 344157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344175 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344214 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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31ee82863c |
Generate response to Status Enquiry message with Status q.931 message.
Some PBXes require this for call status checking
(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
........
Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 344159 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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c1647ab33a |
Fix pin parameter behavior regression in MeetMe
The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5203e0f552 |
don't call ltohl() twice on the same value
ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344049 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9537f22c54 |
Residual changes for Asterisk v10 branch from ASTERISK-18747.
Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ ........ Merged revisions 344004 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bc370b5462 |
Fix crash when dialplan remove include is called with too few arguments.
"dialplan remove include x from y" crashed when the amount of arguments was less than 6. (closes issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: Andrey Solovyev ........ Merged revisions 343936 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343944 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5cb719acec |
Merged revisions 343900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cee432c9d8 |
Fixed reference to incorrect variable if unknown host configured crash.
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 343852 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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a46f6d142c |
Fix boo-boo in prep_tarball script.
A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. ........ Merged revisions 343789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5249a147b8 |
Make "sip show settings" CLI command get RPID flags from the right global page
The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) ........ Merged revisions 343743 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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02f886b5a2 |
Allow built in variables to be used with dynamic weights.
You can now use the built in variables , , and
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.
(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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2b6ebcb9e9 |
respect case changes in peer names on sip reload
ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343691 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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98e494d4a0 |
Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid.
Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343677 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1c526d3d7d |
Prevent BLF subscriptions from causing deadlocks
Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343635 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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00a522c000 |
Correct the default udptl port range.
The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 ........ Merged revisions 343580 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7a5f6684f0 |
Fix deadlock if peer is destroyed while sending MWI notice.
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343578 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f3c03ac1ec |
list all of the codecs associated with a particular format id for CLI command "core show codec"
AST-699 ........ Merged revisions 343533 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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816dc295c2 |
Formatting and doxygen improvements
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343492 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |