[asterisk-dev] call dropped on transfer (sip) bugid 8064

[Adam Goryachev]

Could somebody please take a look at http://bugs.digium.com/view.php?id=8064

This bug seems to have been introduced in asterisk 1.2.9 and has been 
there ever since. If a call arrives on a zap channel, then goes to a 
queue and then a agent and then a local channel and finally a sip channel.

If that destination sip channel transfers the call to another sip 
channel, all channels are dropped.

See the bug for more details please... let me know if I can do anything 
more to help.

Regards,
Adam