25 Apr
2006
25 Apr
'06
2 p.m.
Klaus Darilion schrieb:
this is quit difficult: Which SIP phones? Which
version of Asterisk? ...
I use snom 360 and 200 phones, Asterisk 1.2.7.1 and OpenSER 1.0.1
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.
Of course Asterisk is able to call A or B in the same way.
Regards
Bastian
regards
klaus
Bastian Schern wrote:
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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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