[asterisk-bugs] [Asterisk 0019102]: audio dropped on attended transfer if first call uses g722

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=19102 
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Reported By:                foxfire
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19102
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.17.2 
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:             2011-04-12 04:23 CDT
Last Modified:              2011-04-12 04:23 CDT
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Summary:                    audio dropped on attended transfer if first call
uses g722
Description: 
I consider this to be medium Severity.
If g722 is not used on an attended transfer no problem everything works
fine.
But if g722 is used in the first call there is a problem.
let us take for example user A,B and C.
A calls B ( A uses g722 and B does not have g722 )
B starts the attended transfer and dials to C
while it is ringing and C hasn't answered i get the following warning:

[Apr 11 09:40:58] NOTICE[9507]: channel.c:3137 __ast_read: Dropping
incompatible voice frame on Local/3001 at enumtrans-1628;2 of format g722
since our native format has changed to 0x4 (ulaw)

but no problems in audio just the anoying message constantly in the
console.

C answeres and no more warning messages all is Ok.

Finaly B does a hangup to transfer the call and the real problem apears
audio only in one direction and the following error message:

[Apr 11 09:40:58] NOTICE[9507]: channel.c:3137 __ast_read: Dropping
incompatible voice frame on Local/3001 at enumtrans-1628;2 of format g722
since our native format has changed to 0x4 (ulaw)

i have noticed that when using the Local channel there are some problems i
have tried to avoid local channels in the dialplan , but in this case of an
attended transfer i can't avoid it. By the way i used a Grandstream as A
and Polycom as B and C.

Please let me know which logs you require.
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Issue History 
Date Modified    Username       Field                    Change               
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2011-04-12 04:23 foxfire        New Issue                                    
2011-04-12 04:23 foxfire        Asterisk Version          => 1.6.2.17.2      
2011-04-12 04:23 foxfire        Regression                => No              
2011-04-12 04:23 foxfire        SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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