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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                     acknowledged
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                       SWP-442 
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:              2011-04-13 01:02 CDT
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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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Relationships       ID      Summary
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duplicate of        0017364 atxfer, one way sound, codecs
has duplicate       0017400 Writeformat Slin instead Alaw after att...
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 (0133694) tallguy74 (reporter) - 2011-04-13 01:02
 https://issues.asterisk.org/view.php?id=16287#c133694 
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Is anybody still working on this?

I'm also having this issue, and it's still in asterisk version 1.6.2.17.2
(reproduced yesterday 12 april 2011).
To get around this issue i had to go back to asterisk 1.4 (1.4.40), since
this seems to be broken in every asterisk 1.6 version so far.

I have multiple SIP connection to different PBX's, and use Asterisk as the
proxy for the internal call routing.

If somebody call's from one of the PBX's (A side) to someone on another
PBX (B side) true the asterisk proxy, everything works.
If the B side forwards the call to another PBX( C-side) true asterisk and
does a blind transfer, it works. However, if you will for the C side to
pickup, and then transfer the call, A doesn't hear C while C does hear A.

This prevents us from migrating, and i realy need the TCP SIP...... ;-)  

Also, all connections we're tested with one shared codec (ALAW), and also
pressing a key doesn't resolve the issue. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-13 01:02 tallguy74      Note Added: 0133694                          
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