mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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21941 Commits (d7dec4f14fc018d18ed480358f9a265b81b230b9)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
d7dec4f14f |
Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Stefan Schmidt
|
edaf970c38 |
Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Richard Mudgett
|
7d9ba4875b |
Fix calls to ast_get_ip() not initializing the address family.
........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Walter Doekes
|
2ee874178e |
Minor cleanup in chan_sip get_msg_text() function.
In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Walter Doekes
|
b7aee9ebc9 |
Fix ast_str_truncate signedness warning and documentation.
Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kinsey Moore
|
e6ca768081 |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Jordan
|
76394a727f |
Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Terry Wilson
|
6d05a31d9f |
Resume playing existing hold music for cached realtime MOH
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Paul Belanger
|
f59322f724 |
Added support level for new modules
........ Merged revisions 346029 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
2b3e28f88c |
Fix dnsmgr entries to ask for the same address family each time.
The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Walter Doekes
|
d777e15792 |
Clarify why the AST_LOG_* macros exist next to the LOG_* macros.
(issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Paul Belanger
|
51ce2669af |
Add missing sound_only_one config variable
(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
conf_config_parser.diff (license #5755) patch uploaded by zvision
........
Merged revisions 345882 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
14 years ago |
|
Terry Wilson
|
32d0faac9c |
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Paul Belanger
|
298d015828 |
Add #tryinclude statement
This provides the same functionality as #include however an asterisk module will still load if the filename does not exist. Review: https://reviewboard.asterisk.org/r/1476/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
|
6e7359f594 |
Update the documentation to better clarify how the existing commands work.
Review: https://reviewboard.asterisk.org/r/1593/ ........ Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345683 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Tilghman Lesher
|
6b8b13ec2d |
Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged revisions 345640 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
cd9680e241 |
Accidentally readded sipfriends.sql in r345560. This was removed
in r342871 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
Jonathan Rose
|
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 |
14 years ago |
|
Richard Mudgett
|
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
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14 years ago |
|
Richard Mudgett
|
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
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14 years ago |
|
Richard Mudgett
|
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
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14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
Richard Mudgett
|
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
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14 years ago |
|
Terry Wilson
|
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 |
14 years ago |
|
Walter Doekes
|
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 |
14 years ago |
|
Jonathan Rose
|
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 |
14 years ago |
|
Kinsey Moore
|
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 |
14 years ago |
TransNexus OSP Development
|
f436a6f27c |
Increased max number of destinations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Gregory Nietsky
|
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 |
14 years ago |
|
Terry Wilson
|
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 |
14 years ago |
|
Walter Doekes
|
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 |
14 years ago |
|
Matthew Jordan
|
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 |
14 years ago |
|
Walter Doekes
|
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 |
14 years ago |
|
Walter Doekes
|
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 |
14 years ago |
|
Kinsey Moore
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
Jonathan Rose
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
Richard Mudgett
|
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 |
14 years ago |
|
David Vossel
|
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 |
14 years ago |
|
Kinsey Moore
|
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 |
14 years ago |
|
Kinsey Moore
|
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 |
14 years ago |
Matthew Nicholson
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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 |
14 years ago |
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Richard Mudgett
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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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14 years ago |
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Terry Wilson
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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 |
14 years ago |
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Terry Wilson
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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 |
14 years ago |