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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
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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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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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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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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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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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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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r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines
Merged revisions 281051 via svnmerge from
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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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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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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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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
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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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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.
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Brett Bryant
Richard Mudgett
Tilghman Lesher
Terry Wilson
Olle Johansson
Russell Bryant
Jason Parker
Tzafrir Cohen
David Vossel
Jeff Peeler
Matthew Nicholson
Paul Belanger
Mark Michelson