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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                     acknowledged
Asterisk Version:           1.6.2.17.2 
JIRA:                       SWP-3324 
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:              2011-04-13 03:28 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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 (0133696) foxfire (reporter) - 2011-04-13 03:28
 https://issues.asterisk.org/view.php?id=19102#c133696 
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UPS
the second error is wrong in my description above.i did paste the wrong
line. The error is like found in the log :

Apr 11 09:41:12] WARNING[9529]: chan_sip.c:6290 sip_write: Asked to
transmit frame type 64, while native formats is 0x1000 (g722)(4096)
read/write = 0x40 (slin)(64)/0x1000 (g722)(4096)     

sorry about the mixup 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-13 03:28 foxfire        Note Added: 0133696                          
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