[asterisk-bugs] [Asterisk 0016153]: [patch] Extend slin16 support to SIP calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu May 27 10:50:48 CDT 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=16153
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Reported By: kfister
Assigned To:
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Project: Asterisk
Issue ID: 16153
Category: Core/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.0-rc3
JIRA: SWP-1497
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-10-29 06:59 CDT
Last Modified: 2010-05-27 10:50 CDT
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Summary: [patch] Extend slin16 support to SIP calls
Description:
1.6.2.0-rc3 appears to support using the slin16 codec for IAX2 calls.
These changes extend the functionality to SIP calls. chan_sip must be told
to "allow=slin16."
I have tested this on my own system. It works with Aastra 57i telephone
running firmware 2.5.2.1010.
As of this week SVN 1.6.2 and SVN trunk are also missing this
functionality, similar changes in the relevant files (rtp_engine.c instead
of rtp.c) should implement it.
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(0122569) tilghman (administrator) - 2010-05-27 10:50
https://issues.asterisk.org/view.php?id=16153#c122569
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Patches for new features must be for trunk, not 1.6.2. In addition, slin16
is only meant to be used internally as an intermediate step for transcoding
16kHz audio. Additionally, you've specified the same text description for
slin16 as for slin, which means the codec will never be sourced.
Issue History
Date Modified Username Field Change
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2010-05-27 10:50 tilghman Note Added: 0122569
2010-05-27 10:50 tilghman Status confirmed => feedback
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