[Asterisk-Users] Announced Transfer from Zap to SIP crashes

[James Coberly]

Hi,

If I understand right,  From a Zap station,  you should be able to 
<flashhook> and transfer/3-way a call.  It appears to have issues 
passing calls from PSTN Zap channels -> Zap extension -> SIP.  See below.

When Pushing an inbound PSTN Zap call from a zap answering station,  and 
pressing <Flashhoook> dialing the SIP extension,  SIP stations answers, 
 Press <flashhook> again,  we are now on a threeway call,  when the Zap 
user hangs up,  * loses it /crashes,  with a debug error of File 
Channel.c Line 2252 (ast_channel_bridge) : Nobody there, continuing.
  The SIP user is still on "the call",  the Zap PSTN call is lost, and 
the Zap answering station is hung up.

Is this an issue,  or am I going about this the wrong way?

Thanks in advance.