[asterisk-bugs] [Asterisk 0016153]: Extend slin16 support to SIP calls

Thu Oct 29 06:59:16 CDT 2009

The following issue has been SUBMITTED. 
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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:                     new
Asterisk Version:           1.6.2.0-rc3 
JIRA:                        
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:              2009-10-29 06:59 CDT
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Summary:                    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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Issue History 
Date Modified    Username       Field                    Change               
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2009-10-29 06:59 kfister        New Issue                                    
2009-10-29 06:59 kfister        Asterisk Version          => 1.6.2.0-rc3     
2009-10-29 06:59 kfister        Regression                => No              
2009-10-29 06:59 kfister        SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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