[asterisk-bugs] [Asterisk 0016153]: [patch] Extend slin16 support to SIP calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jun 17 13:36:06 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16153
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Reported By: kfister
Assigned To: dvossel
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Project: Asterisk
Issue ID: 16153
Category: Core/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: assigned
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: 2010-06-17 13:36 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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(0123557) svnbot (reporter) - 2010-06-17 13:36
https://issues.asterisk.org/view.php?id=16153#c123557
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Repository: asterisk
Revision: 271261
U trunk/CHANGES
U trunk/main/rtp_engine.c
U trunk/res/res_rtp_asterisk.c
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r271261 | dvossel | 2010-06-17 13:36:06 -0500 (Thu, 17 Jun 2010) | 9 lines
adds support for slin16 in sip
(closes issue https://issues.asterisk.org/view.php?id=16153)
Reported by: kfister
Patches:
16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd
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http://svn.digium.com/view/asterisk?view=rev&revision=271261
Issue History
Date Modified Username Field Change
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2010-06-17 13:36 svnbot Checkin
2010-06-17 13:36 svnbot Note Added: 0123557
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