[Users] Attended Transfer

Bastian Schern ml02 at in-bln.de
Fri Apr 28 20:53:21 CEST 2006


I made an dump of a working scenario (97-96-91.html) and a dump of the 
problematic scenario (PSTN-96-91.html).

Regards
	Bastian

Klaus Darilion schrieb:
> 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


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