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git.ipfire.org Git - thirdparty/asterisk.git/log
Russell Bryant [Tue, 13 Jul 2010 19:15:47 +0000 (19:15 +0000)]
Recorded merge of revisions 276126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010) | 2 lines
Only reset a CDR that exists.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276127
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Russell Bryant [Tue, 13 Jul 2010 19:09:42 +0000 (19:09 +0000)]
Merged revisions 276123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines
Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276124
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Tilghman Lesher [Tue, 13 Jul 2010 19:05:17 +0000 (19:05 +0000)]
Oops, XML documentation fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276122
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Tilghman Lesher [Tue, 13 Jul 2010 19:00:02 +0000 (19:00 +0000)]
It really cannot fail in the places below, but the stupid compiler doesn't know that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276120
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Tilghman Lesher [Tue, 13 Jul 2010 18:41:59 +0000 (18:41 +0000)]
Weird compiler error on Bamboo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276118
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Tilghman Lesher [Tue, 13 Jul 2010 18:31:41 +0000 (18:31 +0000)]
FILE() now supports line-mode and writing (altering) files.
(closes issue #16461)
Reported by: skyman
Patches:
20100622__issue16461.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276114
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Jeff Peeler [Tue, 13 Jul 2010 17:37:40 +0000 (17:37 +0000)]
Merged revisions 275773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines
Make user removals and traversals thread safe in meetme.
Race conditions present in meetme involving the user list where a lack of
locking has the potential for a user to be removed during a traversal or as in
the case of the reporter after checking if the list is empty could cause a
crash. Fixing this was done by convering the userlist to an ao2 container.
(closes issue #17390)
Reported by: Vince
Review: https://reviewboard.asterisk.org/r/746/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074
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Terry Wilson [Tue, 13 Jul 2010 17:11:37 +0000 (17:11 +0000)]
Destroy RTP fds when we schedule final dialog destruction
Since we are only keeping the dialog around for retransmissions at this point
and there is no possibility that we are still handling RTP, go ahead and
destroy the RTP sessions. Keeping them alive for 32 past when they are used
is unnecessary and can lead to problems with having too many open file
descriptors, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998
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Russell Bryant [Tue, 13 Jul 2010 16:53:44 +0000 (16:53 +0000)]
Merged revisions 275994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines
Access peer->cdr directly instead of through a saved off reference.
At this point in the code, it is possible that peer_cdr may be invalid.
Specifically, in the blind transfer code, CDRs are swapped between channels.
So, peer_cdr is no longer == peer->cdr.
The scenario that exposed a crash in this code was a blind transfer that hit
the system call limit, causing the transferee channel to get destroyed after
the transfer attempt failed. Even if it succeeds and this code doesn't crash,
this code was still trying to reset a CDR on a channel that was now owned by
a different thread, which is a BadThing(tm).
(ABE-2417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275995
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Tilghman Lesher [Tue, 13 Jul 2010 14:48:40 +0000 (14:48 +0000)]
Merged revisions 275909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines
Move SQL scripts into their own database-specific directories.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275910
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Russell Bryant [Tue, 13 Jul 2010 11:41:54 +0000 (11:41 +0000)]
Add example script for use with the externpasscheck voicemail.conf option.
(closes issue #17628)
Reported by: lmadsen
Tested by: russell, lmadsen
Review: https://reviewboard.asterisk.org/r/774/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275863
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Terry Wilson [Mon, 12 Jul 2010 23:27:42 +0000 (23:27 +0000)]
Don't try to ref authpeer when it isn't set
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275816
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Richard Mudgett [Mon, 12 Jul 2010 17:54:46 +0000 (17:54 +0000)]
Add which ITU spec specifies the numbering plan.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275725
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Jeff Peeler [Mon, 12 Jul 2010 17:21:01 +0000 (17:21 +0000)]
Merged revisions 275665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines
Change ast_write to not stop generator when called from ast_prod.
For SIP channels configured with the progressinband option on, the ringback was
being immediately stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the generator
triggered the check in ast_write to stop the generator. The fix here should
write the frame the same as was done before the call to ast_prod was moved.
(closes issue #17372)
Reported by: tech_admin
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275682
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Leif Madsen [Mon, 12 Jul 2010 15:37:01 +0000 (15:37 +0000)]
cdr_pgsql does not detect when a table is found.
This change adds an ERROR message to let you know when a failure exists to
get the columns from the pgsql database, which typically means that the
table does not exist.
(closes issue #17478)
Reported by: kobaz
Patches:
cdr_pgsql.patch uploaded by kobaz (license 834)
Tested by: kobaz, russell, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275626
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Mark Michelson [Mon, 12 Jul 2010 14:55:23 +0000 (14:55 +0000)]
Allow netsock2.c to compile on systems that do not define AI_NUMERICSERV.
(closes issue #17617)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17617.patch uploaded by pprindeville (license 347)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275587
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TransNexus OSP Development [Mon, 12 Jul 2010 04:16:18 +0000 (04:16 +0000)]
Added support for indirect work mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275551
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Eliel C. Sardanons [Sat, 10 Jul 2010 20:49:30 +0000 (20:49 +0000)]
When creating a conference for a unit test, it is not mandatory to open a
dahdi pseudo channel, so if we fail doing it, continue creating the conference.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275509
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Russell Bryant [Sat, 10 Jul 2010 14:48:03 +0000 (14:48 +0000)]
Make indentation consistent, move some queue features to the queue section.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275467
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Russell Bryant [Sat, 10 Jul 2010 14:44:18 +0000 (14:44 +0000)]
Add support for devices with less than 3 lines on the LCD.
(closes issue #17600)
Reported by: minaguib
Patches:
ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466
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Russell Bryant [Fri, 9 Jul 2010 21:57:21 +0000 (21:57 +0000)]
Fix some issues related to dynamic feature groups in features.conf.
The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.
Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show".
Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.
Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].
Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.
(closes issue #17589)
Reported by: lmadsen
Patches:
issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275424
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Mark Michelson [Fri, 9 Jul 2010 20:58:52 +0000 (20:58 +0000)]
Fix error in parsing SIP registry strings from ASTdb.
It was essentially an off-by-one error. The easiest way
to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS
macro to parse the pieces of the registration string out. Tested
and it works wonderfully.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275385
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Tilghman Lesher [Fri, 9 Jul 2010 20:01:01 +0000 (20:01 +0000)]
Get more information about the Bamboo test failures
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275312
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Russell Bryant [Fri, 9 Jul 2010 19:58:06 +0000 (19:58 +0000)]
Add missing ao2_iterator_destroy().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275310
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Russell Bryant [Fri, 9 Jul 2010 19:56:41 +0000 (19:56 +0000)]
Fix compile error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275309
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Mark Michelson [Fri, 9 Jul 2010 19:46:20 +0000 (19:46 +0000)]
Fix port parsing in check_via.
If a Via header contained an IPv6 address, we would not properly parse
the port. We would instead get the information after the first colon in
the address.
(closes issue #17614)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275308
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Paul Belanger [Fri, 9 Jul 2010 19:32:47 +0000 (19:32 +0000)]
Include rdnis in msgXXXX.txt file.
(closes issue #17566)
Reported by: outcast
Patches:
voicemail-rdnis.patch uploaded by outcast (license 1071)
Tested by: outcast
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275307
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Mark Michelson [Fri, 9 Jul 2010 19:29:30 +0000 (19:29 +0000)]
Fix an issue where the port for p->ourip was being set to 0.
This should fix all the CDR tests that were not passing. When they would
originate a call, all fields in the INVITE that contained the source port would
have the port set to 0. Most troubling of these was the Contact header. Tests
are passing locally now and should also pass on the bamboo build agents.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275294
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Paul Belanger [Fri, 9 Jul 2010 19:21:27 +0000 (19:21 +0000)]
Merged revisions 275241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines
Fix logging message for stale nonce.
(closes issue #17582)
Reported by: kenner
Patches:
chan_sip.c.diff uploaded by kenner (license 1040)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275249
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Tilghman Lesher [Fri, 9 Jul 2010 18:55:02 +0000 (18:55 +0000)]
Weird, no output and Bamboo still fails...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227
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Matthew Nicholson [Fri, 9 Jul 2010 18:24:03 +0000 (18:24 +0000)]
Merged revisions 275182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines
give a better error message when attempting to unload a module that is not loaded
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275186
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Tilghman Lesher [Fri, 9 Jul 2010 18:21:39 +0000 (18:21 +0000)]
Add some diagnostic feedback to our data tests
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275172
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Russell Bryant [Fri, 9 Jul 2010 18:11:13 +0000 (18:11 +0000)]
Move parking lot sample config out from the middle of dynamic features sample config.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275147
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Matthew Nicholson [Fri, 9 Jul 2010 17:50:45 +0000 (17:50 +0000)]
Merged revisions 275143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines
don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275144
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Tilghman Lesher [Fri, 9 Jul 2010 17:00:22 +0000 (17:00 +0000)]
Kill some startup warnings and errors and make some messages more helpful in tracking down the source.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105
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Mark Michelson [Fri, 9 Jul 2010 16:39:16 +0000 (16:39 +0000)]
Return logic of sip_debug_test_addr() to its original functionality.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275104
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Matthew Nicholson [Fri, 9 Jul 2010 16:05:58 +0000 (16:05 +0000)]
Merged revisions 275027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines
Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(closes issue #17592)
Reported by: jamicque
Patches:
G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: jamicque, mnicholson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275028
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Russell Bryant [Fri, 9 Jul 2010 15:35:53 +0000 (15:35 +0000)]
Merged revisions 275021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275022
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Mark Michelson [Fri, 9 Jul 2010 14:27:07 +0000 (14:27 +0000)]
Fix sip_uri_parse test comparison.
Part of the change with the IPv6 changes is to treat a host:port as
a single 'domain' entity. This test was not updated to have the correct
expectation after calling parse_uri().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984
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<simon.perreault@viagenie.ca> [Fri, 9 Jul 2010 13:30:37 +0000 (13:30 +0000)]
Copy the address into the peer structure after we set the default port
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274947
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<simon.perreault@viagenie.ca> [Fri, 9 Jul 2010 12:56:18 +0000 (12:56 +0000)]
Sadly we can't dereference a pointer cast and use it as an lvalue without getting this
warning (at least with gcc 4.4.4):
netsock2.c:492: warning: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
So we're back to using memcpy()...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274909
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Russell Bryant [Fri, 9 Jul 2010 12:48:25 +0000 (12:48 +0000)]
Extend length limit on country name in indications.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274907
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Olle Johansson [Fri, 9 Jul 2010 11:06:19 +0000 (11:06 +0000)]
Make it possible to disable individual cdr files per accountcode in cdr_csv
Review: https://reviewboard.asterisk.org/r/678/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274866
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Richard Mudgett [Thu, 8 Jul 2010 23:46:20 +0000 (23:46 +0000)]
Fix calls of ast_sockaddr_from_sin() from IPv6 integration.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274828
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Richard Mudgett [Thu, 8 Jul 2010 23:23:17 +0000 (23:23 +0000)]
Fix compile of chan_ooh323.c from IPv6 integration.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274827
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Mark Michelson [Thu, 8 Jul 2010 22:16:16 +0000 (22:16 +0000)]
And the automerge property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274786
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Mark Michelson [Thu, 8 Jul 2010 22:15:25 +0000 (22:15 +0000)]
Delete properties I merged during v6-new merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274785
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Mark Michelson [Thu, 8 Jul 2010 22:08:07 +0000 (22:08 +0000)]
Add IPv6 to Asterisk.
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783
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Richard Mudgett [Thu, 8 Jul 2010 22:05:40 +0000 (22:05 +0000)]
Generate a correct AstData string for ast_callerid.cid_ton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274782
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Richard Mudgett [Thu, 8 Jul 2010 19:12:55 +0000 (19:12 +0000)]
Fix trunk compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274773
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Eliel C. Sardanons [Thu, 8 Jul 2010 14:48:42 +0000 (14:48 +0000)]
Implement AstData API data providers as part of the GSOC 2010 project,
midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727
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David Vossel [Wed, 7 Jul 2010 20:09:00 +0000 (20:09 +0000)]
Fixes some ref count issues introduced by r274539
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274686
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Richard Mudgett [Wed, 7 Jul 2010 18:32:35 +0000 (18:32 +0000)]
Add missing conditional around chan_dahdi mfcr2_skip_category config parameter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274639
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Richard Mudgett [Wed, 7 Jul 2010 18:20:00 +0000 (18:20 +0000)]
Merged revisions 274579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line
Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274595
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Matthew Nicholson [Wed, 7 Jul 2010 16:40:19 +0000 (16:40 +0000)]
Set proper FAXOPT(status), FAXOPT(statusstr), and FAXOPT(error) values where possible. Previously some failure cases did not result in proper FAXOPT values.
FAX-203
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274540
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Mark Michelson [Wed, 7 Jul 2010 16:21:53 +0000 (16:21 +0000)]
Use the relatedpeer field of a sip_pvt during INVITE processing.
Review: https://reviewboard.asterisk.org/r/629
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274539
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TransNexus OSP Development [Wed, 7 Jul 2010 07:07:08 +0000 (07:07 +0000)]
Changed OSP TCP port from 1080 to 5045.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274492
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Tilghman Lesher [Wed, 7 Jul 2010 06:32:39 +0000 (06:32 +0000)]
Also run the externnotify script when the pollmailboxes thread notices a change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274491
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Tilghman Lesher [Wed, 7 Jul 2010 06:15:43 +0000 (06:15 +0000)]
Merged revisions 274417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines
Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(closes issue #16102)
Reported by: Delvar
Patches:
say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274418
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Jeff Peeler [Tue, 6 Jul 2010 22:23:35 +0000 (22:23 +0000)]
Merged revisions 274283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274316
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Terry Wilson [Tue, 6 Jul 2010 22:15:27 +0000 (22:15 +0000)]
Merged revisions 274280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
........
(no option for trunk, just changing the behavior)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284
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Tilghman Lesher [Tue, 6 Jul 2010 22:09:23 +0000 (22:09 +0000)]
Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274281
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Matthew Nicholson [Tue, 6 Jul 2010 19:53:04 +0000 (19:53 +0000)]
Properly detect and report invalid maxrate and maxrate values in the FAXOPT dialplan function. Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.
FAX-202
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274243
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Mark Michelson [Tue, 6 Jul 2010 14:31:13 +0000 (14:31 +0000)]
Merged revisions 274157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274164
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Matthew Nicholson [Tue, 6 Jul 2010 13:52:56 +0000 (13:52 +0000)]
Blocked revisions 274093 via svnmerge
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r274093 | mnicholson | 2010-07-06 08:52:28 -0500 (Tue, 06 Jul 2010) | 2 lines
Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274094
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Tilghman Lesher [Tue, 6 Jul 2010 06:01:37 +0000 (06:01 +0000)]
Uh, yeah.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274053
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Tilghman Lesher [Mon, 5 Jul 2010 20:00:48 +0000 (20:00 +0000)]
Blocked revisions 273981 via svnmerge
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r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines
Command 'stop gracefully' doesn't.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273982
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Paul Belanger [Mon, 5 Jul 2010 13:53:44 +0000 (13:53 +0000)]
Merged revisions 273884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273886
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Tilghman Lesher [Sat, 3 Jul 2010 02:36:31 +0000 (02:36 +0000)]
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830
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Tilghman Lesher [Fri, 2 Jul 2010 17:10:59 +0000 (17:10 +0000)]
Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273718
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:50 +0000 (16:57 +0000)]
Blocked revisions 273639 via svnmerge
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r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines
If all members are paused, the wrong status is indicated.
(closes issue #17576)
Reported by: ramonpeek
Patches:
diff.txt uploaded by ramonpeek (license 266)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273715
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:28 +0000 (16:57 +0000)]
The switch fallthrough could create some errorneous situations, so best to force directly to the default case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714
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Tzafrir Cohen [Fri, 2 Jul 2010 15:57:02 +0000 (15:57 +0000)]
Fix various typos reported by Lintian
(Also fix the typos in the comments)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641
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Russell Bryant [Thu, 1 Jul 2010 22:16:23 +0000 (22:16 +0000)]
Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273566
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Jeff Peeler [Thu, 1 Jul 2010 20:28:15 +0000 (20:28 +0000)]
Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522
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Matthew Nicholson [Thu, 1 Jul 2010 19:34:47 +0000 (19:34 +0000)]
Properly handle failures of fax->start_session()
FAX-177
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273464
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David Vossel [Thu, 1 Jul 2010 16:40:17 +0000 (16:40 +0000)]
correct handling of get_destination return values
A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273427
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Jeff Peeler [Thu, 1 Jul 2010 15:12:31 +0000 (15:12 +0000)]
Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273355
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Matthew Nicholson [Thu, 1 Jul 2010 14:37:37 +0000 (14:37 +0000)]
Fixed whitespace problems
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273352
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Matthew Nicholson [Thu, 1 Jul 2010 14:34:31 +0000 (14:34 +0000)]
Altered my comment about TCP_NODELAY
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273350
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Matthew Nicholson [Thu, 1 Jul 2010 12:57:18 +0000 (12:57 +0000)]
Don't free written frames in chan_mobile's mbl_write() function.
(closes issue #16430)
Reported by: azbest
Tested by: azbest
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273312
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Matthew Nicholson [Wed, 30 Jun 2010 18:48:21 +0000 (18:48 +0000)]
Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This regression was introduced in r48338.
AST-359
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273270
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Paul Belanger [Wed, 30 Jun 2010 17:28:04 +0000 (17:28 +0000)]
Fix rt(c)p set debug ip taking wrong argument
Also clean up some coding errors.
(closes issue #17469)
Reported by: wdoekes
Patches:
astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273233
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Richard Mudgett [Wed, 30 Jun 2010 17:17:05 +0000 (17:17 +0000)]
Remove unnecessary if test in CV_DSTR()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273198
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Richard Mudgett [Wed, 30 Jun 2010 17:15:46 +0000 (17:15 +0000)]
Misc doxygen cleanup in config.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273197
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Tilghman Lesher [Wed, 30 Jun 2010 01:07:02 +0000 (01:07 +0000)]
Permission checking for the system application is backwards.
(closes issue #17550)
Reported by: kenner
Patches:
manager.c.diff uploaded by kenner (license 1040)
Tested by: kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273144
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Tilghman Lesher [Wed, 30 Jun 2010 01:01:14 +0000 (01:01 +0000)]
Don't attempt to proceed if our internal parser indicates an invalid file.
(closes issue #17560)
Reported by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273142
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Tilghman Lesher [Tue, 29 Jun 2010 23:20:40 +0000 (23:20 +0000)]
Merged revisions 273060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all.
(closes issue #16029)
Reported by: Guggemand
Patches:
realtime-useragent.patch uploaded by Guggemand (license 897)
Tested by: Guggemand
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273078
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Tilghman Lesher [Tue, 29 Jun 2010 22:59:51 +0000 (22:59 +0000)]
Recorded merge of revisions 273057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines
_Really_ skip the channel... don't just retry for another 200 cycles.
(Closes issue SWP-1652, ABE-2240)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273058
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Tilghman Lesher [Tue, 29 Jun 2010 22:40:00 +0000 (22:40 +0000)]
Exclude libical for insufficient versions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273055
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Tilghman Lesher [Tue, 29 Jun 2010 22:39:22 +0000 (22:39 +0000)]
Send DialPlanComplete as a response, not as a separate event.
Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273054
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David Vossel [Tue, 29 Jun 2010 20:44:05 +0000 (20:44 +0000)]
send a 400 Bad Request on malformed sip request
RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272981
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Tilghman Lesher [Mon, 28 Jun 2010 21:50:57 +0000 (21:50 +0000)]
Merged revisions 272925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines
Don't change ownership/group/permissions on run directory, if it already exists.
(closes issue #17076)
Reported by: stuarth
Patches:
20100324__issue17076.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272926
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Tilghman Lesher [Mon, 28 Jun 2010 21:42:52 +0000 (21:42 +0000)]
Merged revisions 272921-272922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines
Change the way that we read include files, to accommodate for changes in GCC 4.4.
(closes issue #17472)
Reported by: seandarcy
Patches:
config2.patch uploaded by nivan (license 1066)
Tested by: nivan
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r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines
Also trim trailing blanks on #includes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272923
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David Vossel [Mon, 28 Jun 2010 18:38:47 +0000 (18:38 +0000)]
rfc compliant sip option parsing + new unit test
RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.
This is not currently being done correctly. Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test verifying
this functionality has been created. Some code refactoring was required.
Review: https://reviewboard.asterisk.org/r/680/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272880
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Mark Michelson [Mon, 28 Jun 2010 17:33:12 +0000 (17:33 +0000)]
Merged revisions 272804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines
Decode URI in contact header of 302 response.
ABE-2352
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272805
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Russell Bryant [Mon, 28 Jun 2010 15:33:32 +0000 (15:33 +0000)]
Use the underscore package so that underscores do not need to be escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684
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David Vossel [Mon, 28 Jun 2010 14:55:25 +0000 (14:55 +0000)]
code guidelines cleanup for retrans_pkt() function
I am doing work in this function. I noticed a large number of
coding guidline fixes that needed to be made. Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272652
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Tilghman Lesher [Fri, 25 Jun 2010 20:18:47 +0000 (20:18 +0000)]
Merged revisions 272562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
Make the structure of the table specified before match the queries and results.
(closes issue #17557)
Reported by: cmaj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568
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Matthew Nicholson [Fri, 25 Jun 2010 19:42:54 +0000 (19:42 +0000)]
Implemement support for handling multiple documents when sending.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272558
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projects / thirdparty / asterisk.git / log
