[Users] Attended Transfer

[Klaus Darilion]

this is quit difficult: Which SIP phones? Which version of Asterisk? ...

You have to make sure that Asterisk and the SIP phones are "compatible". 
There are several ways how to make a call transfer.

Also an often seen problem is the different dialing plans on openser and 
Asterisk. Asterisk must be able to call B in the same way (same request 
URI) then A calls B.

regards
klaus

Bastian Schern wrote:
> Hello,
> 
> does anybody got a working configuration to make an "attended call 
> transfer" with a call through an Asterisk gateway?
> 
> Example:
> 
> PSTN --> Asterisk --> SER --+-- A
>                             |
>                             +-- B
> 
> The call will come from the PSTN Network and will go through "A". A sets 
> the call on "Hold" and calls "B". After A is connected with B, A hangup 
> an B got the call from PSTN.
> 
> This in _not_ working at the moment.
> 
> Attended call transfer only with OpenSER and only SIP-Phones is no 
> Problem. But if the is an Asterisk as PSTN-GW in the game it will not work.
> 
> Regards
>     Bastian
> 
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