Jeff Peeler [Mon, 7 Apr 2008 21:35:48 +0000 (21:35 +0000)]
Merged revisions 113013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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Mark Michelson [Mon, 7 Apr 2008 20:22:06 +0000 (20:22 +0000)]
This is a "fix" for something that's been bugging the crap out of me for a while.
The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113207
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Tilghman Lesher [Mon, 7 Apr 2008 19:06:46 +0000 (19:06 +0000)]
Merged revisions 113117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines
Force ast_mktime() to check for DST, since strptime(3) does not.
(Closes issue #12374)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113172
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Terry Wilson [Mon, 7 Apr 2008 18:57:21 +0000 (18:57 +0000)]
atoi(NULL) is bad
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113170
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Jason Parker [Mon, 7 Apr 2008 18:02:51 +0000 (18:02 +0000)]
Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines
Allow playback with noanswer (and add earlyrtp option).
(closes issue #9077)
Reported by: pj
Patches:
earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119
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Mark Michelson [Mon, 7 Apr 2008 16:12:30 +0000 (16:12 +0000)]
Merged revisions 113065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines
This fix prevents a deadlock that was experienced in chan_local. There was
deadlock prevention in place in chan_local, but it would not work in a specific
case because the channel was recursively locked. By unlocking the channel prior
to calling the generator's generate callback in ast_read_generator_actions(), we
prevent the recursive locking, and therefore the deadlock.
(closes issue #12307)
Reported by: callguy
Patches:
12307.patch uploaded by putnopvut (license 60)
Tested by: callguy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113066
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Jeff Peeler [Mon, 7 Apr 2008 15:18:10 +0000 (15:18 +0000)]
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013
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Joshua Colp [Mon, 7 Apr 2008 14:54:42 +0000 (14:54 +0000)]
Put my slinfactory changes back in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113009
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Tilghman Lesher [Sat, 5 Apr 2008 13:24:12 +0000 (13:24 +0000)]
AsyncAGI should not close the manager session on error.
(closes issue #12370)
Reported by: srt
Patches:
asterisk-12370.diff uploaded by srt (license 378)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112972
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Terry Wilson [Sat, 5 Apr 2008 07:58:42 +0000 (07:58 +0000)]
Clean up some memory leak/ref counting issues
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112939
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Terry Wilson [Sat, 5 Apr 2008 04:59:25 +0000 (04:59 +0000)]
Multi-line support for phoneprov
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112906
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Steve Murphy [Sat, 5 Apr 2008 01:33:13 +0000 (01:33 +0000)]
Found a little problem with the sip request handling that could lead to a quick crash of asterisk, and a road to a DOS attack if left unfixed.
Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.
I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short.
Instead of freeing the struct and nulling the pointer, it now just resets it, because this
ast_str is expected by the calling routine to still be there after handle_request_do() returns.
This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem.
I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well;
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874
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Philippe Sultan [Fri, 4 Apr 2008 19:28:49 +0000 (19:28 +0000)]
Merged revisions 112820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line
Free newly allocated channel before returning
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821
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Philippe Sultan [Fri, 4 Apr 2008 17:32:46 +0000 (17:32 +0000)]
Merged revisions 112766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines
Prevent call connections when codecs don't match.
(closes issue #10604)
Reported by: keepitcool
Patches:
branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785
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Dwayne M. Hubbard [Fri, 4 Apr 2008 00:57:33 +0000 (00:57 +0000)]
sleep long enough for the zaptel timer error message to display before exit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112714
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Joshua Colp [Fri, 4 Apr 2008 00:53:19 +0000 (00:53 +0000)]
Merged revisions 112711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines
Pass in the path to Zaptel for systems that install Zaptel headers in a separate location.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112712
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Joshua Colp [Fri, 4 Apr 2008 00:49:01 +0000 (00:49 +0000)]
Blocked revisions 112709 via svnmerge
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r112709 | file | 2008-04-03 21:48:01 -0300 (Thu, 03 Apr 2008) | 2 lines
One thing at a time... let's get 1.4 building.
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Dwayne M. Hubbard [Fri, 4 Apr 2008 00:32:05 +0000 (00:32 +0000)]
blocked for trunk....woot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112708
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Dwayne M. Hubbard [Thu, 3 Apr 2008 22:19:43 +0000 (22:19 +0000)]
satisfy buildbot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112656
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Dwayne M. Hubbard [Thu, 3 Apr 2008 22:13:11 +0000 (22:13 +0000)]
add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112653
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Mark Michelson [Thu, 3 Apr 2008 14:35:47 +0000 (14:35 +0000)]
Merged revisions 112599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines
Fix the testing of the "res" variable so that it is more logically correct and
makes the correct warning and debug messages print.
(closes issue #12361)
Reported by: one47
Patches:
chan_zap_deferred_digit.patch uploaded by one47 (license 23)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112600
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Tilghman Lesher [Thu, 3 Apr 2008 07:49:05 +0000 (07:49 +0000)]
Use a 32k file buffer on recordings, which increases the efficiency of file recording.
(closes issue #11962)
Reported by: garlew
Patches:
recording.patch uploaded by garlew (license 376)
bug-11962.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112564
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Tilghman Lesher [Wed, 2 Apr 2008 19:34:52 +0000 (19:34 +0000)]
Make MISDN generate channel rename events when the name changes.
(closes issue #11142)
Reported by: julianjm
Patches:
chan_misdn_tmpchan_trunk_v1.diff uploaded by julianjm (license 99)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112520
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Mark Michelson [Wed, 2 Apr 2008 17:36:49 +0000 (17:36 +0000)]
Merged revisions 112468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines
Fix a race condition in the manager. It is possible that a new manager event
could be appended during a brief time when the manager is not waiting for input.
If an event comes during this period, we need to set an indicator that there is an
event pending so that the manager doesn't attempt to wait forever for an event that
already happened.
(closes issue #12354)
Reported by: bamby
Patches:
manager_race_condition.diff uploaded by bamby (license 430)
(comments added by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112469
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Joshua Colp [Wed, 2 Apr 2008 15:26:51 +0000 (15:26 +0000)]
Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off.
(closes issue #12169)
Reported by: pj
Patches:
12169-debugoff-2.diff uploaded by qwell (license 4)
Tested by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112431
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Terry Wilson [Wed, 2 Apr 2008 15:25:48 +0000 (15:25 +0000)]
Re-add HTTP post support by moving to res_http_post.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112426
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Mark Michelson [Wed, 2 Apr 2008 14:32:43 +0000 (14:32 +0000)]
Merged revisions 112393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr 2008) | 6 lines
Ensure that there is no timeout if none is specified.
(closes issue #12349)
Reported by: johnlange
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112394
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Jeff Peeler [Tue, 1 Apr 2008 22:55:28 +0000 (22:55 +0000)]
Added dnsmgr status output for sip show registry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112360
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Steve Murphy [Tue, 1 Apr 2008 22:45:10 +0000 (22:45 +0000)]
Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112357
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Russell Bryant [Tue, 1 Apr 2008 22:25:45 +0000 (22:25 +0000)]
Fix a typo that prevented configuration of non-dynamic peers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112351
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Jeff Peeler [Tue, 1 Apr 2008 22:07:30 +0000 (22:07 +0000)]
Existing DNS manager lookups extended to check for SRV records.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112321
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Steve Murphy [Tue, 1 Apr 2008 20:02:19 +0000 (20:02 +0000)]
(closes issue #12298)
Reported by: falves11
Patches:
12298.patch1 uploaded by murf (license 17)
Tested by: murf
I have hopes that the changes made over the last few days will
finalize and solidify this code. While there are bound to be
small tweaks still needed, I feel that the job (at last) is
somewhat completed. Finally, I had a chance to comprehend how
the scoring of extension patterns was done in the previous
version, and I've come very close to using the exact same
criteria in the new pattern matching code. The left-right
sorting is now replicated in the trie structure itself, such
that the first match found will the 'best' match. Compared
the results against 1.4 for several extensions. Replicated
falves11's setup and it works. Used some devious patterns
provided by jsmith, supplemented with a few of my own.
Looks good.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112289
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Jason Parker [Tue, 1 Apr 2008 18:27:08 +0000 (18:27 +0000)]
Minor formatting cleanup.
(closes issue #12343)
Reported by: travishein
Patches:
app_voicemail_code_convention.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112252
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Jason Parker [Tue, 1 Apr 2008 18:24:56 +0000 (18:24 +0000)]
More voicemail doxygen additions/cleanup.
(issue #12343)
Reported by: travishein
Patches:
app_voicemail_code_documentation.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112241
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Tilghman Lesher [Tue, 1 Apr 2008 18:23:40 +0000 (18:23 +0000)]
Fix last commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112234
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Joshua Colp [Tue, 1 Apr 2008 18:06:13 +0000 (18:06 +0000)]
Merged revisions 112209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines
Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things.
(closes issue #12212)
Reported by: bamby
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112210
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Jeff Peeler [Tue, 1 Apr 2008 17:53:08 +0000 (17:53 +0000)]
This adds DNS SRV record support to DNS manager. If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207
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Joshua Colp [Tue, 1 Apr 2008 17:48:52 +0000 (17:48 +0000)]
Merged revisions 112204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines
Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered.
(closes issue #11823)
Reported by: SDamm
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112205
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Joshua Colp [Tue, 1 Apr 2008 17:24:45 +0000 (17:24 +0000)]
Demote a log message down to a warning.
(closes issue #12345)
Reported by: caio1982
Patches:
limit_msg.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112155
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Mark Michelson [Tue, 1 Apr 2008 17:23:19 +0000 (17:23 +0000)]
Merged revisions 112138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines
Initialize the __res_state structure used for dns purposes
to all 0's prior to using it. This is due to valgrind's complaints
on issue #12284 as well as an excerpt found in "Description" portion
of the online man page found here:
http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV
(pertains to issue #12284 but does not necessarily close it)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112148
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Joshua Colp [Tue, 1 Apr 2008 16:50:37 +0000 (16:50 +0000)]
Merged revisions 112125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines
Ensure that we do not exceed the hold's maximum size with a single frame.
(closes issue #12047)
Reported by: fabianoheringer
Tested by: fabianoheringer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112126
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Russell Bryant [Tue, 1 Apr 2008 16:35:04 +0000 (16:35 +0000)]
Now that zaptel trunk has been removed, add the PSTN deprecation notice to chan_zap, as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112124
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Jason Parker [Mon, 31 Mar 2008 22:16:34 +0000 (22:16 +0000)]
I missed a place when this define was changed.
(closes issue #12334)
Reported by: ovi
Patches:
12334-asterisk.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112071
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Jason Parker [Mon, 31 Mar 2008 21:48:30 +0000 (21:48 +0000)]
Merged revisions 112068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | 5 lines
Fix a silly infinite loop when choosing an invalid option.
(closes issue #12315)
Reported by: jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112069
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Terry Wilson [Mon, 31 Mar 2008 21:01:59 +0000 (21:01 +0000)]
Yeah, simplify that logic a bit...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112035
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Terry Wilson [Mon, 31 Mar 2008 20:45:05 +0000 (20:45 +0000)]
Handle blank prefix= in http.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112033
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Russell Bryant [Mon, 31 Mar 2008 17:14:58 +0000 (17:14 +0000)]
Ensure configure gets run on a clean checkout.
(closes issue #12197)
Reported by: juggie
Patches:
12197.diff uploaded by juggie (license 24)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111998
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Russell Bryant [Mon, 31 Mar 2008 16:37:13 +0000 (16:37 +0000)]
This fixes a high fence violation that MALLOC_DEBUG reported to me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111996
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Joshua Colp [Mon, 31 Mar 2008 14:20:39 +0000 (14:20 +0000)]
Initialize all these here tmp pointers at declaration. They confused some compilers a wee bit.
(closes issue #12333)
Reported by: ovi
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111961
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Russell Bryant [Fri, 28 Mar 2008 22:50:46 +0000 (22:50 +0000)]
Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111909
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Russell Bryant [Fri, 28 Mar 2008 22:45:43 +0000 (22:45 +0000)]
Note a minor race condition that I noticed while reviewing Jeff's changes
to this code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111908
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Jason Parker [Fri, 28 Mar 2008 21:46:02 +0000 (21:46 +0000)]
Merged revisions 111856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines
Allow gsm to compile correctly on x86 with gcc4 optimizations.
(closes issue #11243)
Reported by: whiskerp
Patches:
11243-maybe-asm.diff uploaded by qwell (license 4)
Tested by: Seggy (IRC)
Note: While I did write this patch, I would not have found this if fossil
had not reported and fixed issue #12253. A huge thanks to him for helping
to (indirectly) find the problem here.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111857
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Mark Michelson [Fri, 28 Mar 2008 20:03:16 +0000 (20:03 +0000)]
This time the fix is proper for issue 12284. I have tested it thoroughly and found
that valgrind no longer complains and that calls do complete correctly.
The fix is along the same lines as before: Make sure the final null terminator gets copied
into the new sip_request's data pointer. Without it, parse_request will read and potentially
write past the end of the string, causing potential crashes.
(closes issue #12284...for real this time!)
reported by falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111811
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Mark Michelson [Fri, 28 Mar 2008 19:14:51 +0000 (19:14 +0000)]
Temporary revert of 111662. It's causing lots of trouble and appears to not be
the proper solution to the problem reported anyway.
(related to issue #12884)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111777
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Jason Parker [Fri, 28 Mar 2008 19:08:35 +0000 (19:08 +0000)]
Replace magic number size from msgArray array with a define.
(same patch as before, I just split this part out)
(close issue #12326)
Reported by: travishein
Patches:
app_voicemail_code_documentation.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111774
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Jason Parker [Fri, 28 Mar 2008 19:06:54 +0000 (19:06 +0000)]
Add a bit of doxygen documentation for app_voicemail.
(issue #12326)
Reported by: travishein
Patches:
app_voicemail_code_documentation.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111773
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Jason Parker [Fri, 28 Mar 2008 17:57:12 +0000 (17:57 +0000)]
Merged revisions 111720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line
Remove unimplemented softkeys. Prompted by issue #12325.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111721
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Mark Michelson [Fri, 28 Mar 2008 16:36:59 +0000 (16:36 +0000)]
The copy_request function did not take into account the necessary null terminator
for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.
(closes issue #12284)
Reported by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662
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Jason Parker [Fri, 28 Mar 2008 16:20:59 +0000 (16:20 +0000)]
Merged revisions 111658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines
The file size of WAV49 does not need to be an even number.
(closes issue #12128)
Reported by: mdu113
Patches:
12128-noevenlength.diff uploaded by qwell (license 4)
Tested by: qwell, mdu113
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111659
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Tilghman Lesher [Fri, 28 Mar 2008 14:37:28 +0000 (14:37 +0000)]
Merged revisions 111605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines
Update debugging text, since Valgrind eliminated the --log-file-exactly option.
(Closes issue #12320)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111606
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Joshua Colp [Fri, 28 Mar 2008 00:55:47 +0000 (00:55 +0000)]
Forgetting to unregister a manager action is bad, mmmk?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111565
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Mark Michelson [Fri, 28 Mar 2008 00:12:52 +0000 (00:12 +0000)]
Fix a crash that would happen when attempting to unload the app_queue module.
The problem was that when the refcount on the queue hit 0, the destructor was
called, and inside the destructor, another function was called which would increase
the refcount back to 1 again and then decrease it again back to 0 for every member
in the queue. This meant that the destructor was being recursively called, leading
to a double free of the queue. This is now fixed by making sure to unlink the
queue from the queues container prior to the final unref of the queue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111533
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Terry Wilson [Thu, 27 Mar 2008 22:10:25 +0000 (22:10 +0000)]
Fix another little http problem. In making it match coding guidelines, a comparison was dropped
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111500
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Steve Murphy [Thu, 27 Mar 2008 21:25:55 +0000 (21:25 +0000)]
comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111497
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Tilghman Lesher [Thu, 27 Mar 2008 19:26:45 +0000 (19:26 +0000)]
Merged revisions 111442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines
For FreeBSD, at least, the ifa_addr element could be NULL.
(closes issue #12300)
Reported by: festr
Patches:
acl.c.patch uploaded by festr (license 443)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111443
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Steve Murphy [Thu, 27 Mar 2008 13:29:41 +0000 (13:29 +0000)]
Merged revisions 111391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines
These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111410
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Steve Murphy [Thu, 27 Mar 2008 04:47:12 +0000 (04:47 +0000)]
Merged revisions 111341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines
(closes issue #12302)
Reported by: pj
Tested by: murf
These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111360
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Jason Parker [Thu, 27 Mar 2008 00:27:35 +0000 (00:27 +0000)]
But we can change the API here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111295
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Jason Parker [Thu, 27 Mar 2008 00:25:56 +0000 (00:25 +0000)]
Merged revisions 111280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line
Put this flag back so we don't change the API.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111285
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Jason Parker [Wed, 26 Mar 2008 23:27:33 +0000 (23:27 +0000)]
Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines
Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.
While it would be very nice to keep this as optimized as possible, it makes no sense
for the smoother to be dropping random bits of audio like this. Isn't that the
whole point of a smoother?
Closes issue #12093.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111246
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Terry Wilson [Wed, 26 Mar 2008 21:23:29 +0000 (21:23 +0000)]
Stupid strcasecmp function :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111213
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Tilghman Lesher [Wed, 26 Mar 2008 20:34:05 +0000 (20:34 +0000)]
Oops, missed one
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111185
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Tilghman Lesher [Wed, 26 Mar 2008 19:58:09 +0000 (19:58 +0000)]
Simplify new macro, simplify configfile logic, now that list is sorted
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111132
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Joshua Colp [Wed, 26 Mar 2008 19:56:40 +0000 (19:56 +0000)]
Merged revisions 111129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines
Update autosupport script.
(closes issue #12310)
Reported by: angler
Patches:
autosupport.diff uploaded by angler (license 106)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111130
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Kevin P. Fleming [Wed, 26 Mar 2008 19:52:27 +0000 (19:52 +0000)]
Merged revisions 111126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 111125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines
update UPGRADE notes to document usage of the script
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111127
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Mark Michelson [Wed, 26 Mar 2008 19:39:23 +0000 (19:39 +0000)]
Merged revisions 111121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines
This code change is made just for clarification. It does exactly
the same thing as before. It just doesn't look as wrong.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111123
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Joshua Colp [Wed, 26 Mar 2008 19:29:26 +0000 (19:29 +0000)]
Add expiry value to the sip show subscriptions CLI command.
(closes issue #12025)
Reported by: agx
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111083
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Mark Michelson [Wed, 26 Mar 2008 19:26:23 +0000 (19:26 +0000)]
Merged revisions 111049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines
Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.
(closes issue #10487)
Reported by: ewilhelmsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111067
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Tilghman Lesher [Wed, 26 Mar 2008 19:19:31 +0000 (19:19 +0000)]
Add a linkedlist macro that maintains a sorted list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111036
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Jason Parker [Wed, 26 Mar 2008 19:16:31 +0000 (19:16 +0000)]
Only try to detect silence when we actually need to, instead of...always.
If this is wrong, I'd love to hear why.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111028
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Kevin P. Fleming [Wed, 26 Mar 2008 19:08:00 +0000 (19:08 +0000)]
Merged revisions 111024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 111019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines
add a script to make getting the iLBC source code simple for end users
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Jason Parker [Wed, 26 Mar 2008 19:05:51 +0000 (19:05 +0000)]
Large cleanup of DSP code
Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.
2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.
3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.
4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.
5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.
(closes issue #11968)
Reported by: dimas
Patches:
v2-11968-dsp.patch uploaded by dimas (license 88)
v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022
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Joshua Colp [Wed, 26 Mar 2008 19:05:42 +0000 (19:05 +0000)]
Merged revisions 111020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines
If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be.
(closes issue #11995)
Reported by: fall
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111021
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Joshua Colp [Wed, 26 Mar 2008 18:42:52 +0000 (18:42 +0000)]
Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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Tilghman Lesher [Wed, 26 Mar 2008 18:41:27 +0000 (18:41 +0000)]
Oops, fix this, too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111013
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Tilghman Lesher [Wed, 26 Mar 2008 18:39:06 +0000 (18:39 +0000)]
Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012
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Kevin P. Fleming [Wed, 26 Mar 2008 17:44:09 +0000 (17:44 +0000)]
Merged revisions 110962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines
add note that the user will need to enable codec_ilbc to get it to build
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110963
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Donny Kavanagh [Wed, 26 Mar 2008 17:28:49 +0000 (17:28 +0000)]
revert something dumb, because i was running svn diff in a subfolder not the root of trunk, before doing my commit and did not see it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110930
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Donny Kavanagh [Wed, 26 Mar 2008 17:24:54 +0000 (17:24 +0000)]
update documentation to reflect the changes in the way configure detects net-snmp.
(closes issue #12067)
Reported by: juggie
Patches:
12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110911
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Kevin P. Fleming [Wed, 26 Mar 2008 17:10:28 +0000 (17:10 +0000)]
Merged revisions 110880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines
due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves
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Mark Michelson [Wed, 26 Mar 2008 00:02:31 +0000 (00:02 +0000)]
This ensures that the manager interface is not enabled by default. Prior to this
change, it was possible to start Asterisk with the manager interface enabled, then
either comment out the enabled option or make manager.conf unopenable and the manager
interface would still be enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110831
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Jason Parker [Tue, 25 Mar 2008 22:51:55 +0000 (22:51 +0000)]
Merged revisions 110779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines
Make file access in cdr_custom similar to cdr_csv.
Fixes issue #12268.
Patch borrowed from r82344
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110780
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Jeff Peeler [Tue, 25 Mar 2008 20:02:57 +0000 (20:02 +0000)]
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110726
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Tilghman Lesher [Tue, 25 Mar 2008 17:46:34 +0000 (17:46 +0000)]
Update sample configurations to make virtual hosting more obvious.
(closes issue #11969)
Reported by: pprindeville
Patches:
acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110691
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Tilghman Lesher [Tue, 25 Mar 2008 17:40:28 +0000 (17:40 +0000)]
Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293)
Reported by: pprindeville
Patches:
bugid-0012293.1.6.patch uploaded by pprindeville (license 347)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110689
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Mark Michelson [Tue, 25 Mar 2008 15:44:01 +0000 (15:44 +0000)]
Oops here too. I need to stop coding for a while...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110639
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Mark Michelson [Tue, 25 Mar 2008 15:41:33 +0000 (15:41 +0000)]
Merged revisions 110635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines
When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.
So this is a revert of a revert...sort of.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110636
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Joshua Colp [Tue, 25 Mar 2008 15:18:41 +0000 (15:18 +0000)]
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631
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Joshua Colp [Tue, 25 Mar 2008 14:39:45 +0000 (14:39 +0000)]
Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629
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Olle Johansson [Tue, 25 Mar 2008 10:54:07 +0000 (10:54 +0000)]
Use the "Server" header when responding to SIP requests.
(closes issue #12278)
Reported by: rjain
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110625
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