[asterisk-users] Understand SIP REFER

[Johansson Olle E]

19 feb 2009 kl. 00.08 skrev Klaus Darilion:

> Hi!
>
> I have some problems understanding the concept of REFER in Asterisk  
> 1.4.23.
>
> I have the following scenario:
>
> Incoming SIP call (incoming leg) from a SIP trunk into Asterisk  
> (handled
> in context fromTrunk), forwarded to the SIP Client (outgoing leg).
>
> Now, the SIP Client sends a REFER request (unattended transfer) to
> another extension. This terminates the outgoing leg and the incoming  
> leg
> continues dialplan processing in the context of the SIP client
> (fromSipClient).
>
> Processing in the client's context is IMO fine, but the problem is  
> that
> the channel is the incoming channel from the trunk. So the  
> fromSipClient
> is processed by the trunk channel. This in my case does not work as it
> expects to have certain variables set (setvar in sip.conf) - but these
> variables are not present as the new extension is executed by the
> trunk's channel.
>
> I it possible to execute the second call setup completely in the SIP
> clients settings (e.g. loading the clients setvar options)?

I would suggest you do some processing in the TRANSFER_CONTEXT
and load the variables you need. The SIPPEER dialplan function
might be useful.

/O