Terry Wilson [Mon, 16 Aug 2010 20:40:55 +0000 (20:40 +0000)]
Merged revisions 282468 via svnmerge from
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r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
Merged revisions 282467 via svnmerge from
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r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
Merged revisions 282430 via svnmerge from
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r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially
changing, so send a SRCCHANGE so that RTP-based media streams can get
a new SSRC generated to reflect the change. Original patch by addix
(along with lots of testing--thanks!).
(closes issue #17007)
Reported by: addix
Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
srcchange.diff uploaded by twilson (license 396)
Tested by: addix, twilson
Review: https://reviewboard.asterisk.org/r/862/
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Leif Madsen [Mon, 16 Aug 2010 18:02:29 +0000 (18:02 +0000)]
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r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
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r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
Add information about creating sounds files using
the sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds releases.
This allows people creating their own prompts to sound consistent with
the prompts available from the open source project.
SWP-595
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Tzafrir Cohen [Sun, 15 Aug 2010 13:08:45 +0000 (13:08 +0000)]
Support for GNU/kFreeBSD
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See
http://glibc-bsd.alioth.debian.org/porting/PORTING
This patch gets Asterisk close to building on Debian kFreeBSD i386,
mainly by adding an extra test for __GLIBC__ in one or two (or more)
places.
OSARCH is set to 'kfreebsd-gnu'
DAHDI support (and support for chan_vpb) was not tested.
Review: https://reviewboard.asterisk.org/r/858/
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Tilghman Lesher [Sat, 14 Aug 2010 04:58:34 +0000 (04:58 +0000)]
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r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/
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Richard Mudgett [Fri, 13 Aug 2010 23:57:21 +0000 (23:57 +0000)]
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r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially
requested, the saved original dial string was not transferred to the new B
channel. CCSS uses that dial string to generate the recall dial string.
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David Vossel [Fri, 13 Aug 2010 22:27:20 +0000 (22:27 +0000)]
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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David Vossel [Fri, 13 Aug 2010 20:12:22 +0000 (20:12 +0000)]
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r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
res_stun_monitor and corresponding options CHANGES documentation
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David Vossel [Fri, 13 Aug 2010 20:05:44 +0000 (20:05 +0000)]
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r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
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David Vossel [Fri, 13 Aug 2010 18:58:49 +0000 (18:58 +0000)]
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r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
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r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
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Terry Wilson [Fri, 13 Aug 2010 16:08:05 +0000 (16:08 +0000)]
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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Jason Parker [Thu, 12 Aug 2010 22:52:33 +0000 (22:52 +0000)]
Merged revisions 282131 via svnmerge from
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r282131 | qwell | 2010-08-12 17:51:44 -0500 (Thu, 12 Aug 2010) | 16 lines
Merged revisions 282130 via svnmerge from
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r282130 | qwell | 2010-08-12 17:50:54 -0500 (Thu, 12 Aug 2010) | 9 lines
Merged revisions 282129 via svnmerge from
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r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line
Register CLI commands before parsing config, in case there is a config error.
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Richard Mudgett [Thu, 12 Aug 2010 22:10:49 +0000 (22:10 +0000)]
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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Russell Bryant [Thu, 12 Aug 2010 20:44:39 +0000 (20:44 +0000)]
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r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
Add a "core reload" CLI command.
Review: https://reviewboard.asterisk.org/r/859/
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David Vossel [Thu, 12 Aug 2010 20:17:17 +0000 (20:17 +0000)]
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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Russell Bryant [Thu, 12 Aug 2010 18:04:19 +0000 (18:04 +0000)]
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r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) | 2 lines
Put back pointer value output for ast_debug(), such that it is only removed for verbose output.
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Russell Bryant [Thu, 12 Aug 2010 16:48:54 +0000 (16:48 +0000)]
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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Jeff Peeler [Thu, 12 Aug 2010 03:08:45 +0000 (03:08 +0000)]
Merged revisions 281913 via svnmerge from
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r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines
Merged revisions 281912 via svnmerge from
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r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines
Merged revisions 281911 via svnmerge from
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r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged,
which is what used to happen. It seems that fixes were made to attempt limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change.
There are two other control frames sent in ast_channel_bridge that probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave
this change up to the discretion of resolving issue #17007.
For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540
(closes issue #17404)
Reported by: sdolloff
Patches:
bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff
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Leif Madsen [Wed, 11 Aug 2010 21:13:06 +0000 (21:13 +0000)]
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r281875 | lmadsen | 2010-08-11 16:12:13 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281873 via svnmerge from
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r281873 | lmadsen | 2010-08-11 16:09:47 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281819 via svnmerge from
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r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines
Add Danish support to say.conf.sample
(closes issue #17836)
Reported by: RoadKill
Patches:
say.conf.sample.patch.dk uploaded by RoadKill (license 933)
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Matthew Nicholson [Wed, 11 Aug 2010 21:12:25 +0000 (21:12 +0000)]
Merged revisions 281874 via svnmerge from
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r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines
handle all possible responses to REFER requests
(closes issue #17486)
Reported by: davidw
Patches:
Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
Tested by: davidw
Review: https://reviewboard.asterisk.org/r/837/
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Richard Mudgett [Wed, 11 Aug 2010 20:38:57 +0000 (20:38 +0000)]
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r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines
Fix a call to analog_set_pulsedial() not setting 0 or 1 only.
* Also a couple minor tweaks.
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Leif Madsen [Wed, 11 Aug 2010 17:55:45 +0000 (17:55 +0000)]
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r281764 | lmadsen | 2010-08-11 12:54:56 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281763 via svnmerge from
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r281763 | lmadsen | 2010-08-11 12:54:09 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281762 via svnmerge from
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r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines
Allow say.conf to handle large numbers ending with multiple zeros.
(closes issue #17833)
Reported by: RoadKill
Patches:
say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933)
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Matthew Nicholson [Wed, 11 Aug 2010 17:29:16 +0000 (17:29 +0000)]
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r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug 2010) | 4 lines
Avoid a deadlock in add_header_max_forwards().
Related to r276951
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Tilghman Lesher [Wed, 11 Aug 2010 15:20:29 +0000 (15:20 +0000)]
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r281723 | tilghman | 2010-08-11 10:18:40 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281722 via svnmerge from
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r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) | 7 lines
Only set status TIMEOUT, if we have no digits.
(closes issue #15188)
Reported by: jcovert
Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551)
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<simon.perreault@viagenie.ca> [Wed, 11 Aug 2010 13:31:39 +0000 (13:31 +0000)]
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r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines
Fix parsing of IPv6 address literals in outboundproxy
(closes issue #17757)
Reported by: oej
Patches:
17757.diff uploaded by sperreault (license 252)
sip.conf.diff uploaded by sperreault (license 252)
Tested by: oej
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Russell Bryant [Tue, 10 Aug 2010 21:50:24 +0000 (21:50 +0000)]
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r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines
Change the default value for alwaysauthreject in sip.conf to "yes".
(closes issue #17756)
Reported by: oej
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Russell Bryant [Tue, 10 Aug 2010 18:05:40 +0000 (18:05 +0000)]
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r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines
Merged revisions 281574 via svnmerge from
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r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines
Don't move the time threshold for running scheduled events on every iteration.
Instead, only calculate the time threshold each time ast_sched_runq() is called.
(closes issue #17742)
Reported by: schmidts
Patches:
sched.c.patch uploaded by schmidts (license 1077)
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Russell Bryant [Tue, 10 Aug 2010 17:49:36 +0000 (17:49 +0000)]
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r281568 | russell | 2010-08-10 12:48:42 -0500 (Tue, 10 Aug 2010) | 22 lines
Merged revisions 281567 via svnmerge from
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r281567 | russell | 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) | 15 lines
Merged revisions 281566 via svnmerge from
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r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines
Reset visible indication after answer.
(closes issue #17641)
Reported by: klaus3000
Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65)
Tested by: schmidts
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Russell Bryant [Tue, 10 Aug 2010 16:55:50 +0000 (16:55 +0000)]
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r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) | 8 lines
Ensure that the proper external address is used for the RTP destination.
(closes issue #17044)
Reported by: ebroad
Tested by: ebroad
Review: https://reviewboard.asterisk.org/r/566/
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Russell Bryant [Tue, 10 Aug 2010 16:22:58 +0000 (16:22 +0000)]
Merged revisions 281529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) | 8 lines
Resolve a problem with channel name tab completion.
Hitting tab without typing any part of a channel name resulted in no results.
This now results in getting a full list of active channels, just as it did
in previous versions of Asterisk.
Review: https://reviewboard.asterisk.org/r/818/
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TransNexus OSP Development [Tue, 10 Aug 2010 07:26:59 +0000 (07:26 +0000)]
Fixed the issue caused by EXTEN including user parameters.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281498
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Jeff Peeler [Mon, 9 Aug 2010 23:04:59 +0000 (23:04 +0000)]
Merged revisions 281466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09 Aug 2010) | 2 lines
Add some more stuff to copy from 281429.
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David Vossel [Mon, 9 Aug 2010 20:49:13 +0000 (20:49 +0000)]
Merged revisions 281432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281432 | dvossel | 2010-08-09 15:47:53 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
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Jeff Peeler [Mon, 9 Aug 2010 20:46:50 +0000 (20:46 +0000)]
Merged revisions 281429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281429 | jpeeler | 2010-08-09 15:43:54 -0500 (Mon, 09 Aug 2010) | 27 lines
Merged revisions 281391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local
channel would cause the information to be lost. The problem was that the
information was set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and if so
preswap the Caller ID data so that basically the masquerade puts the data back.
(closes issue #17138)
Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/
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Matthew Nicholson [Mon, 9 Aug 2010 14:52:21 +0000 (14:52 +0000)]
Merged revisions 281358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug 2010) | 4 lines
Validate minrate, maxrate, and modem settings before attempting a fax session.
FAX-224
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<simon.perreault@viagenie.ca> [Mon, 9 Aug 2010 14:32:59 +0000 (14:32 +0000)]
Merged revisions 281356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 (Mon, 09 Aug 2010) | 2 lines
Added comment about IPv4-mapped IPv6 addresses and the output of netstat.
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Russell Bryant [Mon, 9 Aug 2010 12:52:04 +0000 (12:52 +0000)]
Merged revisions 281325 via svnmerge from
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r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09 Aug 2010) | 2 lines
Add a couple of default values to the documentation of cdr.conf.
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Russell Bryant [Mon, 9 Aug 2010 12:14:58 +0000 (12:14 +0000)]
Merged revisions 281294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 Aug 2010) | 5 lines
Reorder some options in cdr.conf.sample.
Put all of the options that affect the contents of CDRs together, instead
of having the batch mode options in the middle of them.
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Damien Wedhorn [Sat, 7 Aug 2010 22:36:08 +0000 (22:36 +0000)]
Fix up handling and indications during transfer.
Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up.
(closes issue #17692)
Reported by: jmhunter
Patches:
chan_skinny-transfer-v4.txt uploaded by DEA (license 3)
skinnytransfver.v8.diff uploaded by wedhorn (license 30)
Tested by: jmhunter, salecha, wedhorn
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281257
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Damien Wedhorn [Sat, 7 Aug 2010 22:17:10 +0000 (22:17 +0000)]
Move call answering stuff into new setsubstate_connected.
Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected.
(closes issue #17772)
Reported by: wedhorn
Patches:
cleanup.stateconnected2.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281227
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Damien Wedhorn [Sat, 7 Aug 2010 22:07:43 +0000 (22:07 +0000)]
Start rtp on answer before the answer is queued
(closes issue #17770)
Reported by: salecha
Patches:
skinny.answercrash.diff uploaded by wedhorn (license 30)
Tested by: salecha
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281226
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Tilghman Lesher [Fri, 6 Aug 2010 18:58:39 +0000 (18:58 +0000)]
Merged revisions 281085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) | 8 lines
Fix alignment of stringfields on the SPARC architecture
(closes issue #17789)
Reported by: Ian Mason
Patches:
20100806__issue17789__2.diff.txt uploaded by tilghman (license 14)
Tested by: Ian_Mason
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Russell Bryant [Thu, 5 Aug 2010 13:19:52 +0000 (13:19 +0000)]
Merged revisions 281052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines
Merged revisions 281051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists.
That is no longer the case.
Apply a default value to the unanswered option.
Define all default values as named constants.
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Tilghman Lesher [Thu, 5 Aug 2010 07:47:30 +0000 (07:47 +0000)]
Merged revisions 280984 via svnmerge from
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r280984 | tilghman | 2010-08-05 02:46:36 -0500 (Thu, 05 Aug 2010) | 22 lines
Merged revisions 280983 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280983 | tilghman | 2010-08-05 02:40:47 -0500 (Thu, 05 Aug 2010) | 15 lines
Merged revisions 280982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines
Change context lock back to a mutex, because functionality depends upon the lock being recursive.
(closes issue #17643)
Reported by: zerohalo
Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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Matthew Nicholson [Wed, 4 Aug 2010 15:22:02 +0000 (15:22 +0000)]
Merged revisions 280909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug 2010) | 2 lines
Initialize FAXOPT() status variables in sendfax and receivefax instead of when the details structure is created.
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Tilghman Lesher [Wed, 4 Aug 2010 14:05:11 +0000 (14:05 +0000)]
Merged revisions 280879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 Aug 2010) | 14 lines
Check cur value before attempting a deref.
(closes issue #17775)
Reported by: svinson
Patches:
20100804__issue17775.diff.txt uploaded by tilghman (license 14)
Tested by: svinson
(closes issue #17743)
Reported by: tgruenberg
Patches:
20100804__issue17775.diff.txt uploaded by tilghman (license 14)
Tested by: tgruenberg
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Tilghman Lesher [Tue, 3 Aug 2010 20:29:51 +0000 (20:29 +0000)]
Merged revisions 280809 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines
Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item.
Matches up with FIELDQTY() and CUT().
(closes issue #17713)
Reported by: gareth
Patches:
svn-279754.diff uploaded by gareth (license 208)
Tested by: gareth, tilghman
Review: https://reviewboard.asterisk.org/r/810/
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<simon.perreault@viagenie.ca> [Tue, 3 Aug 2010 19:59:37 +0000 (19:59 +0000)]
Merged revisions 280778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 Aug 2010) | 9 lines
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
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<simon.perreault@viagenie.ca> [Tue, 3 Aug 2010 19:58:59 +0000 (19:58 +0000)]
Merged revisions 280777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 (Tue, 03 Aug 2010) | 8 lines
Better documentation related to IPv6.
(closes issue #17737)
Reported by: oej
Patches:
doc.diff uploaded by sperreault (license 252)
Tested by: mmichelson
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<simon.perreault@viagenie.ca> [Tue, 3 Aug 2010 19:05:50 +0000 (19:05 +0000)]
Reverted r280706 and r280707. Will commit in branch 1.8 and merge to trunk properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280745
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Russell Bryant [Tue, 3 Aug 2010 18:50:14 +0000 (18:50 +0000)]
Merged revisions 280742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) | 4 lines
Remove the MP3 decoder source code and replace it with a small shell script.
Review: https://reviewboard.asterisk.org/r/836/
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Tilghman Lesher [Tue, 3 Aug 2010 18:43:37 +0000 (18:43 +0000)]
Merged revisions 280740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines
Merged revisions 280739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines
Document -B and -W flags and regenerate manpage from sgml
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<simon.perreault@viagenie.ca> [Tue, 3 Aug 2010 16:52:01 +0000 (16:52 +0000)]
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280707
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<simon.perreault@viagenie.ca> [Tue, 3 Aug 2010 16:34:36 +0000 (16:34 +0000)]
Better documentation related to IPv6.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280706
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Tilghman Lesher [Mon, 2 Aug 2010 21:28:09 +0000 (21:28 +0000)]
Merged revisions 280672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280672 | tilghman | 2010-08-02 16:27:25 -0500 (Mon, 02 Aug 2010) | 9 lines
Merged revisions 280671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines
Allow the pipe, but also allow the comma
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Tilghman Lesher [Mon, 2 Aug 2010 14:42:38 +0000 (14:42 +0000)]
Merged revisions 280628 via svnmerge from
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r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010) | 2 lines
Make this a little more deterministic... we want the latest value, not just a 1 somewhere.
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David Vossel [Mon, 2 Aug 2010 14:30:50 +0000 (14:30 +0000)]
if totag is not present for an ACK request, do not send an error response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280627
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Tilghman Lesher [Mon, 2 Aug 2010 14:28:29 +0000 (14:28 +0000)]
Merged revisions 280624 via svnmerge from
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r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010) | 2 lines
Apparently, the values in makeopts are sometimes 1:1 and sometimes 1. Compensate for this.
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Damien Wedhorn [Fri, 30 Jul 2010 09:12:55 +0000 (09:12 +0000)]
Cleanup transmit_ for handle_register and keepalives
Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session).
(closes issue #16980)
Reported by: wedhorn
Patches:
skinny-clean06b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, DEA
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280589
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Matthew Nicholson [Thu, 29 Jul 2010 21:08:42 +0000 (21:08 +0000)]
Merged revisions 280557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul 2010) | 4 lines
Fix regression introduced in r1664. Give the fax stack time to shutdown and populate the FAXOPT output variables.
FAX-222
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Paul Belanger [Thu, 29 Jul 2010 21:06:13 +0000 (21:06 +0000)]
PeerStatus now includes Address and Port
(closes issue #17730)
Reported by: jkroon
Patches:
iax2-peerstate-address.patch uploaded by jkroon (license 714)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280555
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David Vossel [Thu, 29 Jul 2010 20:44:37 +0000 (20:44 +0000)]
Merged revisions 280552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280552 | dvossel | 2010-07-29 15:43:47 -0500 (Thu, 29 Jul 2010) | 17 lines
Merged revisions 280551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines
fixes wrong SRV query for TLS connection
(closes issue #17612)
Reported by: marcelloceschia
Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079)
chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079)
Tested by: marcelloceschia, st, pabelanger
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Russell Bryant [Thu, 29 Jul 2010 20:36:18 +0000 (20:36 +0000)]
Merged revisions 280549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280549 | russell | 2010-07-29 15:35:30 -0500 (Thu, 29 Jul 2010) | 5 lines
Add header to ccss.conf to appease oej.
(closes issue #17755)
Reported by: oej
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Sean Bright [Thu, 29 Jul 2010 19:48:38 +0000 (19:48 +0000)]
Merged revisions 280519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul 2010) | 7 lines
Fix compilation error in chan_dahdi (strdupa -> ast_strdupa).
(closes issue #17751)
Reported by: b11d
Patches:
strdupa_oops.diff uploaded by malcolmd (license 924)
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David Vossel [Thu, 29 Jul 2010 19:35:34 +0000 (19:35 +0000)]
respond with 481 when request requiring totag has no totag to match against
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280518
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David Vossel [Thu, 29 Jul 2010 19:18:50 +0000 (19:18 +0000)]
Merged revisions 280450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
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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)]
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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)]
Merged revisions 280307 via svnmerge from
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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)]
Merged revisions 280235 via svnmerge from
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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)]
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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)]
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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
Merged revisions 279946 via svnmerge from
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r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
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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)]
Merged revisions 279755 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279602
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/1.6.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279600
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279569
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
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