[asterisk-bugs] [Asterisk 0016236]: Can't initiate outbound SIP calls when siren14 is the ONLY enabled codec

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16236 
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Reported By:                ttbrowning
Assigned To:                tilghman
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Project:                    Asterisk
Issue ID:                   16236
Category:                   Core/CodecInterface
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-397 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 229670 
Request Review:              
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Date Submitted:             2009-11-12 14:43 CST
Last Modified:              2009-12-03 12:13 CST
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Summary:                    Can't initiate outbound SIP calls when siren14 is
the ONLY enabled codec
Description: 
Using either a spool file or CLI commands, Asterisk will never intiate an
outbound SIP call when siren14 is the ONLY codec enabled.  Asterisk
ultimately complains that: [Nov 12 14:59:27] WARNING[21376]:
chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call
to foo

Inbound calls work fine.  Attempts to make outbound calls never send an
INVITE.
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 (0114669) tilghman (administrator) - 2009-12-03 12:13
 https://issues.asterisk.org/view.php?id=16236#c114669 
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The problem here is that you're using the call generator for the call, and
the native format for that is signed linear (slin).  Since there is no
codec translator available for siren14, Asterisk cannot make that
translation, which is why it eliminates siren14 from the list of possible
codecs.  Since your outbound only allows siren14, there are no codecs left
for Asterisk to offer and accomplish a call.  This is why it refuses to
offer a call.

At the current time, the ONLY way you can use siren14 is in a passthrough
call (that is, where Asterisk makes no translation).  This will probably
change in the future, but that is the state currently. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-03 12:13 tilghman       Note Added: 0114669                          
2009-12-03 12:13 tilghman       Status                   assigned => feedback
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