[Users] Attended Transfer

[Bastian Schern]

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:
>> 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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