David Vossel [Thu, 29 Jul 2010 19:18:50 +0000 (19:18 +0000)]
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r280450 | dvossel | 2010-07-29 14:13:27 -0500 (Thu, 29 Jul 2010) | 25 lines
Merged revisions 280449 via svnmerge from
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r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines
Merged revisions 280448 via svnmerge from
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r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines
fixes issue with translator frame not getting freed
A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
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Paul Belanger [Thu, 29 Jul 2010 18:51:39 +0000 (18:51 +0000)]
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r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul 2010) | 2 lines
Remove res_crypto dependency.
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Jean Galarneau [Thu, 29 Jul 2010 16:47:23 +0000 (16:47 +0000)]
Merged revisions 280346 via svnmerge from
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r280346 | jeang | 2010-07-29 11:07:16 -0500 (Thu, 29 Jul 2010) | 17 lines
Merged revisions 280345 via svnmerge from
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r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines
Merged revisions 280341 via svnmerge from
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r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away.
ABE-2422
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Paul Belanger [Thu, 29 Jul 2010 16:45:34 +0000 (16:45 +0000)]
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r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul 2010) | 2 lines
crypto_loaded_test depends on res_crypto, else test will fail.
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Russell Bryant [Thu, 29 Jul 2010 16:26:13 +0000 (16:26 +0000)]
Merged revisions 280391 via svnmerge from
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r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010) | 2 lines
Don't blow up if get_codec() was not provided in the RTP glue.
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Matthew Nicholson [Thu, 29 Jul 2010 15:58:39 +0000 (15:58 +0000)]
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r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu, 29 Jul 2010) | 4 lines
Use PRIx64 instead of PRId64 in format string.
related to r280302
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Matthew Nicholson [Thu, 29 Jul 2010 15:56:26 +0000 (15:56 +0000)]
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r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed, 28 Jul 2010) | 2 lines
Use PRId64 with format_t
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Matthew Nicholson [Thu, 29 Jul 2010 15:41:27 +0000 (15:41 +0000)]
Make chan_usbradio.c build on 64bit platforms.
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Matthew Nicholson [Thu, 29 Jul 2010 14:03:59 +0000 (14:03 +0000)]
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r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines
Merged revisions 280306 via svnmerge from
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r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines
Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229
Review: https://reviewboard.asterisk.org/r/813/
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Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699.
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Jeff Peeler [Wed, 28 Jul 2010 20:50:02 +0000 (20:50 +0000)]
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r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) | 2 lines
Give test category missing leading slash
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Richard Mudgett [Wed, 28 Jul 2010 20:19:21 +0000 (20:19 +0000)]
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r280235 | rmudgett | 2010-07-28 15:12:16 -0500 (Wed, 28 Jul 2010) | 9 lines
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r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines
Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options.
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Jason Parker [Wed, 28 Jul 2010 20:04:06 +0000 (20:04 +0000)]
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r280233 | qwell | 2010-07-28 15:03:22 -0500 (Wed, 28 Jul 2010) | 13 lines
Merged revisions 280231 via svnmerge from
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r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | 6 lines
Work around some silly behavior on BSD.
A non-zero exit from a subshell should make the build fail.
(closes issue #17621)
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Terry Wilson [Wed, 28 Jul 2010 19:37:45 +0000 (19:37 +0000)]
Merged revisions 280225 via svnmerge from
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r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28 Jul 2010) | 3 lines
Do rtp/rtcp debugging when it is turned on w/o filtering
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Jason Parker [Wed, 28 Jul 2010 18:25:00 +0000 (18:25 +0000)]
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r280195 | qwell | 2010-07-28 13:24:29 -0500 (Wed, 28 Jul 2010) | 16 lines
Merged revisions 280193 via svnmerge from
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r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | 9 lines
Remove unnecessary subshells. Attempt to make checksumming work.
Also improves readability.
(issue #17621)
Reported by: bjm
Review: https://reviewboard.asterisk.org/r/808/
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Sean Bright [Wed, 28 Jul 2010 16:53:14 +0000 (16:53 +0000)]
Merged revisions 280161 via svnmerge from
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r280161 | seanbright | 2010-07-28 12:52:12 -0400 (Wed, 28 Jul 2010) | 15 lines
Merged revisions 280160 via svnmerge from
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r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738)
Reported by: bobwienholt
Patches:
issue17738.patch uploaded by bobwienholt (license 950)
Tested by: bobwienholt, seanbright
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Olle Johansson [Wed, 28 Jul 2010 14:14:06 +0000 (14:14 +0000)]
Formatting changes
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Leif Madsen [Wed, 28 Jul 2010 13:53:23 +0000 (13:53 +0000)]
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r280090 | lmadsen | 2010-07-28 08:52:50 -0500 (Wed, 28 Jul 2010) | 16 lines
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r280089 | lmadsen | 2010-07-28 08:51:16 -0500 (Wed, 28 Jul 2010) | 9 lines
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r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line
Update help text to be less confusing.
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Russell Bryant [Wed, 28 Jul 2010 13:02:11 +0000 (13:02 +0000)]
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r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010) | 2 lines
s/init keys/keys init/
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Paul Belanger [Wed, 28 Jul 2010 01:39:29 +0000 (01:39 +0000)]
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r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue, 27 Jul 2010) | 5 lines
Resolve compiler warning about formatting
(closes issue #17732)
Reported by: pabelanger
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Russell Bryant [Tue, 27 Jul 2010 21:16:39 +0000 (21:16 +0000)]
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r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) | 5 lines
Add --enable-coverage option to configure script.
This option enables the proper compiler flags for tracking code coverage, which
is useful along side automated testing.
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David Vossel [Tue, 27 Jul 2010 20:59:16 +0000 (20:59 +0000)]
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r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines
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r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
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r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
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Russell Bryant [Tue, 27 Jul 2010 19:55:40 +0000 (19:55 +0000)]
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r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines
Fix inband DTMF detection on outgoing ISDN calls.
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
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Mark Michelson [Tue, 27 Jul 2010 18:55:06 +0000 (18:55 +0000)]
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r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines
Fix parsing error in sip_sipredirect().
The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
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Jason Parker [Tue, 27 Jul 2010 16:41:19 +0000 (16:41 +0000)]
Merged revisions 279850 via svnmerge from
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r279850 | qwell | 2010-07-27 11:40:05 -0500 (Tue, 27 Jul 2010) | 9 lines
Merged revisions 279849 via svnmerge from
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r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) | 1 line
Simply sounds/Makefile some more.
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David Vossel [Tue, 27 Jul 2010 16:11:11 +0000 (16:11 +0000)]
Merged revisions 279817 via svnmerge from
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r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) | 2 lines
fix sip transaction match with authentication, fix confusing log message when using getaddrinfo
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Russell Bryant [Tue, 27 Jul 2010 16:08:10 +0000 (16:08 +0000)]
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r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines
Ignore a control subclass of -1 in ast_waitfordigit_full().
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r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines
Support "channels" in addition to "channel" in chan_dahdi.conf.
Review: https://reviewboard.asterisk.org/r/804
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Mark Michelson [Tue, 27 Jul 2010 15:16:45 +0000 (15:16 +0000)]
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r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines
Merged revisions 279784 via svnmerge from
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r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
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Paul Belanger [Tue, 27 Jul 2010 03:02:33 +0000 (03:02 +0000)]
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r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, 26 Jul 2010) | 10 lines
If dringXcontext is null, fallback to default context value.
(closes issue #17693)
Reported by: iasgoscouk
Patches:
issue17693.patch uploaded by pabelanger (license 224)
Tested by: iasgoscouk
Review: https://reviewboard.asterisk.org/r/803/
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Paul Belanger [Tue, 27 Jul 2010 01:56:30 +0000 (01:56 +0000)]
Merged revisions 279726 via svnmerge from
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r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul 2010) | 9 lines
Use ast_sockaddr_setnull() when http is not enabled.
Otherwise, ast_tcptls_server_start() will still start http.
(closes issue #17708)
Reported by: pabelanger
Patches:
http.patch uploaded by pabelanger (license 224)
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Russell Bryant [Tue, 27 Jul 2010 01:39:58 +0000 (01:39 +0000)]
Make a formatting change. (Demonstrating the commit IRC bot to pabelanger)
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Paul Belanger [Mon, 26 Jul 2010 23:35:03 +0000 (23:35 +0000)]
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r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines
Updated documentation for FAX logger level.
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Jason Parker [Mon, 26 Jul 2010 23:06:47 +0000 (23:06 +0000)]
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r279658 | qwell | 2010-07-26 18:03:38 -0500 (Mon, 26 Jul 2010) | 12 lines
Merged revisions 279657 via svnmerge from
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r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines
Really fix sounds Makefile (and make it readableish).
There was a rather large syntax error that should have caused ALL versions of GNU make to fail.
I don't know how it worked.
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Tilghman Lesher [Mon, 26 Jul 2010 21:21:23 +0000 (21:21 +0000)]
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r279619 | tilghman | 2010-07-26 16:20:12 -0500 (Mon, 26 Jul 2010) | 9 lines
Merged revisions 279609 via svnmerge from
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r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 Jul 2010) | 2 lines
Dunno why this worked on my machine, but it works better this way.
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Tilghman Lesher [Mon, 26 Jul 2010 21:08:24 +0000 (21:08 +0000)]
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r279601 | tilghman | 2010-07-26 16:07:45 -0500 (Mon, 26 Jul 2010) | 19 lines
Merged revisions 279597 via svnmerge from
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r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573
(closes issue #13573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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Gavin Henry [Mon, 26 Jul 2010 21:07:38 +0000 (21:07 +0000)]
Merged revisions 279598 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279598 | ghenry | 2010-07-26 21:58:12 +0100 (Mon, 26 Jul 2010) | 21 lines
Merged revisions 279597 via svnmerge from
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r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573 [^]
(closes issue 0013573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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David Vossel [Mon, 26 Jul 2010 20:00:52 +0000 (20:00 +0000)]
Merged revisions 279568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines
transaction matching using top most Via header
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
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Paul Belanger [Mon, 26 Jul 2010 19:58:12 +0000 (19:58 +0000)]
Merged revisions 279566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines
Add documentation for FAX logger level.
(closes issue #17715)
Reported by: vrban
Patches:
17715.patch uploaded by pabelanger (license 224)
Tested by: vrban
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Tilghman Lesher [Mon, 26 Jul 2010 19:20:23 +0000 (19:20 +0000)]
Merged revisions 279562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279562 | tilghman | 2010-07-26 14:18:26 -0500 (Mon, 26 Jul 2010) | 9 lines
Merged revisions 279561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines
Use a special Makefile for noobs who still have GNU Make 3.80.
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Mark Michelson [Mon, 26 Jul 2010 16:44:25 +0000 (16:44 +0000)]
Merged revisions 279504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
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Sean Bright [Mon, 26 Jul 2010 15:44:46 +0000 (15:44 +0000)]
Merged revisions 279502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279502 | seanbright | 2010-07-26 11:43:54 -0400 (Mon, 26 Jul 2010) | 12 lines
Merged revisions 279501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul 2010) | 5 lines
Expand the correct value within AST_OPTION_ONLY.
(closes issue #17703)
Reported by: stuarth
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Tilghman Lesher [Mon, 26 Jul 2010 03:28:02 +0000 (03:28 +0000)]
Merged revisions 279472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) | 2 lines
Formats need to load before apps, because some apps call ast_format_str_reduce() at load time.
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Paul Belanger [Sun, 25 Jul 2010 21:28:10 +0000 (21:28 +0000)]
Merged revisions 279442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun, 25 Jul 2010) | 2 lines
Add trailing backslash to silence warning message.
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Tilghman Lesher [Sun, 25 Jul 2010 18:22:13 +0000 (18:22 +0000)]
Merged revisions 279410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279410 | tilghman | 2010-07-25 13:21:27 -0500 (Sun, 25 Jul 2010) | 8 lines
Don't re-register CDR module on reload.
(closes issue #17304)
Reported by: jnemeth
Patches:
20100507__issue17304.diff.txt uploaded by tilghman (license 14)
Tested by: jnemeth
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279413
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Tilghman Lesher [Sun, 25 Jul 2010 17:33:45 +0000 (17:33 +0000)]
Merged revisions 279390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010) | 8 lines
Don't assume qlog is open.
(closes issue #17704)
Reported by: vrban
Patches:
issue17704.patch uploaded by pabelanger (license 224)
Tested by: vrban
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279391
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Paul Belanger [Sat, 24 Jul 2010 20:49:39 +0000 (20:49 +0000)]
Merged revisions 279314 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279314 | pabelanger | 2010-07-24 16:47:52 -0400 (Sat, 24 Jul 2010) | 7 lines
Remove duplicate -c flag when using $(INSTALL)
(closes issue #17695)
Reported by: pabelanger
Patches:
Makefile.diff uploaded by pabelanger (license 224)
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Paul Belanger [Sat, 24 Jul 2010 18:20:18 +0000 (18:20 +0000)]
Merged revisions 279280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279280 | pabelanger | 2010-07-24 14:18:43 -0400 (Sat, 24 Jul 2010) | 8 lines
Check if ast_sockaddr is NULL then return.
(closes issue #17677)
Reported by: outcast
Patches:
issue0017677.patch uploaded by pabelanger (license 224)
Tested by: elguero
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Paul Belanger [Sat, 24 Jul 2010 17:54:03 +0000 (17:54 +0000)]
Merged revisions 279273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul 2010) | 6 lines
Default sin_family to AF_INET for TCP / TLS Bindaddress.
Otherwise, 'manager show settings' will generate errors
if manager is not enabled.
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Richard Mudgett [Fri, 23 Jul 2010 22:24:52 +0000 (22:24 +0000)]
Merged revisions 279227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines
Merged revisions 279207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
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Richard Mudgett [Fri, 23 Jul 2010 19:53:03 +0000 (19:53 +0000)]
Make "dahdi show channels" show an outgoing called number.
The "dahdi show channels" extension column previously only showed the
called number of an incoming call. It now shows the called number for an
outgoing call as well.
(closes issue #17653)
Reported by: amazinzay
Patches:
issue17653_trunk.txt uploaded by rmudgett (license 664)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279156
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Russell Bryant [Fri, 23 Jul 2010 19:17:30 +0000 (19:17 +0000)]
Shuffle UPGRADE.txt files for 1.10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118
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Russell Bryant [Fri, 23 Jul 2010 19:16:14 +0000 (19:16 +0000)]
Start a new section in CHANGES for 1.10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279116
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Tilghman Lesher [Fri, 23 Jul 2010 18:56:59 +0000 (18:56 +0000)]
Merged revisions 279113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010) | 2 lines
Silly 64-bit compilers (who uses 64-bit anyway?)
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Russell Bryant [Fri, 23 Jul 2010 18:22:09 +0000 (18:22 +0000)]
Remove old properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279084
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Russell Bryant [Fri, 23 Jul 2010 18:18:25 +0000 (18:18 +0000)]
Add branch-1.8-merged and branch-1.8-blocked properties to trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279063
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Tilghman Lesher [Fri, 23 Jul 2010 17:05:16 +0000 (17:05 +0000)]
Merged revisions 278984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) | 5 lines
Establish a maximum version for openh323 (i.e. not opal), because chan_h323 will fail to load, even if it links.
(issue #17679)
Reported by: am
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Tilghman Lesher [Fri, 23 Jul 2010 16:43:34 +0000 (16:43 +0000)]
Merged revisions 278981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines
Avoid race with consolethread on shutdown (on parallel processors).
(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
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Mark Michelson [Fri, 23 Jul 2010 16:33:52 +0000 (16:33 +0000)]
SIP URI comparison fixes.
This initially was created to work around the issue of
using a string comparison instead of a binary comparison
for IP addresses. It evolved a bit when test cases were
created and it was discovered that comparison of URI
parameters was not working exactly as it should.
sip_uri_cmp() and its helpers have been moved to reqresp_parser.c
and a new test has been added.
(closes issue #17662)
Reported by: oej
Review: https://reviewboard.asterisk.org/r/792
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980
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Tilghman Lesher [Fri, 23 Jul 2010 16:19:21 +0000 (16:19 +0000)]
Merge the realtime failover branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278957
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Tzafrir Cohen [Fri, 23 Jul 2010 16:07:53 +0000 (16:07 +0000)]
Some left-over hyphen-minus fixes in the man page
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278947
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Russell Bryant [Fri, 23 Jul 2010 15:57:23 +0000 (15:57 +0000)]
... just kidding. Enable SIP by default. :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278945
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Russell Bryant [Fri, 23 Jul 2010 15:57:01 +0000 (15:57 +0000)]
Disable SIP support by default for Asterisk 1.8.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278944
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Mark Michelson [Fri, 23 Jul 2010 15:52:37 +0000 (15:52 +0000)]
Well, who knew chan_ooh323 used udptl? I sure didn't!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278943
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Richard Mudgett [Fri, 23 Jul 2010 15:41:44 +0000 (15:41 +0000)]
Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278942
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Mark Michelson [Fri, 23 Jul 2010 15:16:33 +0000 (15:16 +0000)]
Allow IPv6 addresses for UDPTL streams.
Review: https://reviewboard.asterisk.org/r/795
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908
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Olle Johansson [Fri, 23 Jul 2010 13:37:17 +0000 (13:37 +0000)]
Minor corrections to the LDAP realtime driver
Review: https://reviewboard.asterisk.org/r/798/
Thanks Mark for a quick review!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278875
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Paul Belanger [Fri, 23 Jul 2010 13:26:41 +0000 (13:26 +0000)]
Portability updates for Makefiles.
When possible, use $(INSTALL). This allows us to use the functionality within
install for setting directory / file permissions, a requirement for unprivileged
installation.
Also move any directory we plan to create within the installdirs macro. Plus
various other formatting issues.
(issue #17436)
Reported by: pabelanger
Patches:
non-root.patch.v8 uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/654/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278873
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Alec L Davis [Fri, 23 Jul 2010 11:01:14 +0000 (11:01 +0000)]
missed FXS kewl start polarityswitch when finally on hook.
(issue #17318)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278841
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Alec L Davis [Thu, 22 Jul 2010 23:14:50 +0000 (23:14 +0000)]
Support FXS module Polarity Reversal on remote party Answer and Hangup
FXS lines normally connect to a telephone. However, when FXS lines are routed
to an external PBX or Key System to act as "external" or "CO" lines, it is
extremely difficult, if not impossible for the external PBX to know when
the call has been disconnected without receiving a polarity reversal on the line.
Now using answeronpolarityswitch and hanguponpolarityswitch keywords that
previously were used only for FXO ports, now applies like functionality for
an FXS port, but from the connected equipment's point of view.
(closes issue #17318)
Reported by: armeniki
Patches:
fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/797/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278809
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Richard Mudgett [Thu, 22 Jul 2010 21:16:04 +0000 (21:16 +0000)]
DNID not cleared when channel hang up (Affects PRI and SS7)
The "dahdi show channels" CLI command still reports the DNID of the
previous call even if the call is already hang up. The "dahdi show
channels" command of older releases clear the DNID once the channel is
hang up.
Regression from the sig_analog/sig_pri extraction from chan_dahdi.
(closes issue #17623)
Reported by: klaus3000
Patches:
issue17623.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278777
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Jeff Peeler [Thu, 22 Jul 2010 19:45:30 +0000 (19:45 +0000)]
Add method for finding XML doc files for systems that don't support GLOB_BRACE.
In particular, Solaris and perhaps others do not support the above mentioned
GNU extension. In this case the paths are simply expanded without the braces
and the calls to glob are made separately.
Note: I could not explain memory allocation failures that were being reported
from within libxml itself when making calls to glob without using GLOB_NOCHECK.
This is the only reason why that flag is being used.
(closes issue #15402)
Reported by: snuffy
Patches:
bug_xmlpatt-v3.diff uploaded by snuffy (license 35),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278708
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Mark Michelson [Thu, 22 Jul 2010 14:58:01 +0000 (14:58 +0000)]
Merged revisions 278618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines
Allow PLC to function properly when channels use SLIN for audio.
If a channel involved in a bridge was using SLIN audio, then translation
paths were not guaranteed to be set up properly since in all likelihood
the number of translation steps was only 1.
This patch enforces the transcode_via_slin behavior if transcode_via_slin
or generic_plc is enabled and one of the formats to make compatible is
SLIN.
AST-352
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David Vossel [Thu, 22 Jul 2010 14:56:26 +0000 (14:56 +0000)]
update sip subscription debug message to a warning message
If the Expire header of a SUBSCRIBE is less that our expiremin,
a log warning will be displayed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278619
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Tilghman Lesher [Thu, 22 Jul 2010 05:29:29 +0000 (05:29 +0000)]
Add the full current set of CDR drivers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278579
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David Vossel [Wed, 21 Jul 2010 19:16:12 +0000 (19:16 +0000)]
make func_file unit test's category consistent with other tests
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278539
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Terry Wilson [Wed, 21 Jul 2010 19:11:32 +0000 (19:11 +0000)]
Remove built-in AES code and use optional_api instead
Review: https://reviewboard.asterisk.org/r/793/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538
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David Vossel [Wed, 21 Jul 2010 18:52:14 +0000 (18:52 +0000)]
send "423 Interval too small" Response to Subscribe with Expires less that min allowed
[RFC3265]3.1.6.1....
The notifier MAY also check that the duration in the "Expires" header
is not too small. If and only if the expiration interval is greater
than zero AND smaller than one hour AND less than a notifier-
configured minimum, the notifier MAY return a "423 Interval too
small" error which contains a "Min-Expires" header field. The "Min-
Expires" header field is described in SIP [1].
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278536
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Tzafrir Cohen [Wed, 21 Jul 2010 17:44:20 +0000 (17:44 +0000)]
Fix invalid test for rxisoffhook in FXO channels
This fixes some cases of no outgoing calls on FXO before an incoming call.
Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO
(KS/GS) channels.In some cases the bit would not be initialized properly
before the first inbound call and thus prevent an outgoing call.
If those tests are actually required by anybody, they should define
DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c .
(closes issue #14577)
Reported by: jkroon
Patches:
asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by frawd (license 610)
Tested by: frawd
Review: https://reviewboard.asterisk.org/r/699/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278501
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Russell Bryant [Wed, 21 Jul 2010 16:15:00 +0000 (16:15 +0000)]
Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
(closes issue #15915)
Reported by: keiron
(closes issue #17187)
Reported by: Eddie Edwards
(closes issue #16494)
Reported by: Hubguru
(closes issue #15731)
Reported by: flop
(closes issue #12917)
Reported by: falves11
(closes issue #14920)
Reported by: vrban
(closes issue #17199)
Reported by: aleksey2000
(closes issue #15406)
Reported by: kowalma
(closes issue #17438)
Reported by: dcabot
(closes issue #17325)
Reported by: glwgoes
(closes issue #17118)
Reported by: erikje
possibly other issues, too ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278465
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Tilghman Lesher [Wed, 21 Jul 2010 15:56:05 +0000 (15:56 +0000)]
Ensure realtime conferences are treated the same as static conferences when trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime, and add them
to the existing flags.
(closes issue #17502)
Reported by: kenji
Patches:
20100720__issue17502.diff.txt uploaded by tilghman (license 14)
Tested by: kenji
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278463
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Matthew Nicholson [Wed, 21 Jul 2010 15:54:29 +0000 (15:54 +0000)]
Properly show the current page being transfered for 'fax show session'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278462
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Matthew Nicholson [Wed, 21 Jul 2010 15:51:24 +0000 (15:51 +0000)]
Properly set the port number for UDPTL media sessions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278461
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Matthew Nicholson [Wed, 21 Jul 2010 13:03:01 +0000 (13:03 +0000)]
Don't print failure status when the remote end hangs up, it may not be an actual failure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278426
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Russell Bryant [Wed, 21 Jul 2010 13:02:46 +0000 (13:02 +0000)]
Update documentation for 'comebacktoorigin' in featuers.conf.
The documentation for this option did not match the code. Fix that along with
some minor cleanups to the code along the way. Document a slight change in
behavior (to something that was previously undocumented) in UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278425
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Tilghman Lesher [Wed, 21 Jul 2010 06:45:06 +0000 (06:45 +0000)]
Change order so that it more closely matches the related SIP command.
(closes issue #17648)
Reported by: GMLudo
Review: https://reviewboard.asterisk.org/r/789/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278393
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Jeff Peeler [Wed, 21 Jul 2010 03:53:19 +0000 (03:53 +0000)]
include stat.h for everybody, needed for device2chan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278361
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Tilghman Lesher [Tue, 20 Jul 2010 23:23:25 +0000 (23:23 +0000)]
Separate queue_log arguments into separate fields, and allow the text file to be used, even when realtime is used.
(closes issue #17082)
Reported by: coolmig
Patches:
20100720__issue17082.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278307
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Tilghman Lesher [Tue, 20 Jul 2010 22:40:19 +0000 (22:40 +0000)]
Merged revisions 278261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines
Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message.
(closes issue #16350)
Reported by: noahisaac
Patches:
20100623__issue16350.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278275
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Richard Mudgett [Tue, 20 Jul 2010 22:38:13 +0000 (22:38 +0000)]
Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278274
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Tilghman Lesher [Tue, 20 Jul 2010 22:26:23 +0000 (22:26 +0000)]
Merged revisions 278167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278272
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David Vossel [Tue, 20 Jul 2010 21:41:21 +0000 (21:41 +0000)]
fixes sip CANCEL race condition
If Asterisk sends a 4xx error and the other side sends a CANCEl
before receiving the 4xx and responding with the ACK, Asterisk
will process the CANCEL and send a 487 Request Terminated as
a new final response to the INVITE. Since we are issuing a new
final response to the INVITE, the old one must be pretend_acked
else it will keep retransmitting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278234
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Matthew Nicholson [Tue, 20 Jul 2010 21:01:26 +0000 (21:01 +0000)]
This commit contains several changes to the way output channel variables are handled.
FAX output channel variables will now match the values reported by FAXOPT() and should be set in all failure and success cases.
This commit also contains a few modifications to the way FAXOPT() variables are populated in a few spots and fixes for some reference count leaks of the session details structure in some failure cases.
Also found and fixed more cases where FAXOPT(status) may not have gotten set.
FAX-214
FAX-203
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278168
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Tilghman Lesher [Tue, 20 Jul 2010 19:35:02 +0000 (19:35 +0000)]
Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132
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Russell Bryant [Tue, 20 Jul 2010 18:11:08 +0000 (18:11 +0000)]
Add a package to install_prereq.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278096
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Russell Bryant [Tue, 20 Jul 2010 17:22:36 +0000 (17:22 +0000)]
Only call ast_channel_cc_params_init() if allocating a channel succeeds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278051
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Tilghman Lesher [Tue, 20 Jul 2010 16:50:11 +0000 (16:50 +0000)]
Merged revisions 278023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines
Off-by-one error
(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278024
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Jean Galarneau [Mon, 19 Jul 2010 21:07:08 +0000 (21:07 +0000)]
Merged revisions 277906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines
Avoid trying to pickup a parked extension before the park operation is completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
(ABE-2418)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277945
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Mark Michelson [Mon, 19 Jul 2010 17:16:23 +0000 (17:16 +0000)]
Fix port setting of external address in SIP.
There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
(closes issue #17665)
Reported by: mmichelson
Patches:
17665.diff#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873
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Mark Michelson [Mon, 19 Jul 2010 17:10:00 +0000 (17:10 +0000)]
Remove the fe80:1234::1234 test case from test_acl.c
The ACL test was failing on Mac OS X because it would
convert the above invalid link-local address into
fe80::1234 while reporting no error from getaddrinfo().
Linux does not do this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277872
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Jeff Peeler [Mon, 19 Jul 2010 14:39:07 +0000 (14:39 +0000)]
Fix regression with distinctive ring detection.
The issue here is that passing an array to a function prohibits the ARRAY_LEN
macro from returning the real size. To avoid this the size is now defined and
use of ARRAY_LEN is avoided.
(closes issue #15718)
Reported by: alecdavis
Patches:
bug15718.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277837
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Mark Michelson [Mon, 19 Jul 2010 14:17:16 +0000 (14:17 +0000)]
Make ACLs IPv6-capable.
ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814
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