[Kamailio-Users] Attended Transfer with asterisk

[Daniel-Constantin Mierla]

Hello,

On 10/08/08 12:40, ALAEDDINE abbech 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 
>                             |<---transfer call pstn to B 
>                             +-- 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 and B got the call from PSTN. 
> This is not working at the moment. 
> Attended call transfer only with OpenSER and only with SIP-Phones is no Problem. 
> But if the is an Asterisk as PSTN-GW in the game it will not work.
>   
this might be a routing problem or an issue in Asterisk. To be able to 
help more, please post the SIP trace for this case (you can use ngrep to 
catch the sip traffic on kamailio server).

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
http://www.asipto.com