[Users] Attended Transfer

[Klaus Darilion]

This looks fine, except that asterisk does not know how to call the new 
destination: (Refer-To: 
sip:00043551191 at 192.168.0.91:5060;line=fnhf1aep?Replaces=3c267e322bf2-hcvubihsgoqa%40snom360-000413230066%3Bto-tag%3Dgpij52sisv%3Bfrom-tag%3Djhu81hu4vn 
)

watch asterisks log files, make sure asterisk is using the proper 
context when looking in extensions.conf.

google for asterisk transfer context



regards
klaus

Bastian Schern wrote:
> 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
> 
> 
> ____________
> Virus checked by G DATA AntiVirusKit
> Version: AVK 16.7061 from 28.04.2006
> Virus news: www.antiviruslab.com
>