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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