[Users] Attended Transfer

[Klaus Darilion]

then we will need some more SIP dumps to help you.

"ngrep -d any port 5060" on the SIP proxy.

regards
klaus

On Tue, April 25, 2006 20:00, Bastian Schern said:
> 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
>>>
>>> ____________
>>> Virus checked by G DATA AntiVirusKit
>>> Version: AVK 16.7010 from 25.04.2006
>>> Virus news: www.antiviruslab.com
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>
>
> ____________
> Virus checked by G DATA AntiVirusKit
> Version: AVK 16.7010 from 25.04.2006
> Virus news: www.antiviruslab.com
>
>
>