[asterisk-bugs] [Asterisk 0016287]: One way audio after attended transfer

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=16287 
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Reported By:                cbkm
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16287
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.6.2 - please test a newer version 
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:             2009-11-19 11:58 CST
Last Modified:              2009-11-20 08:24 CST
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Summary:                    One way audio after attended transfer
Description: 
Possibly related to 14249 but I couldn't find a way to reopen that bug,
so...

Scenario:-

1. SIP/213 calls SIP/207
2. SIP/207 uses asterisk transfer functionality to transfer to SIP/214
3. SIP/207 hangs up
4. SIP/213 is now bridged to SIP/214
5. Console scrolls with:-

[Nov 19 17:31:59] WARNING[4263] chan_sip.c: Asked to transmit frame type
64, while native formats is 0x4 (ulaw)(4) read/write = 0x8 (alaw)(8)/0x4
(ulaw)(4)

6. If SIP/214 presses a key the message stops scrolling.

Attached is a noisy (verbose 9 / debug 9 / sip debug) log (on the
assumption I can attach a file once I hit submit).

This is with asterisk 1.6.2rc2 (with the patch from issue: 15848) NOT
1.6.2rc6 like it says in the version dropdown - sorry.

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Issue History 
Date Modified    Username       Field                    Change               
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2009-11-20 08:24 lmadsen        Asterisk Version         1.6.2.0-rc6 => Older
1.6.2 - please test a newer version
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