[OpenSIPS-Users] Opensips and Asterisk - Problem with extensions and SIP messages

[Bogdan-Andrei Iancu]

Hi,

could you please draw a small flow of the call, just to understand it... 
Like
    UA(104) ----(INVITE)---> Asterisk -> proxy1 ->....etc

For INVITE, 3xx , etc

Regards,
Bogdan

Jennifer-4 wrote:
> Hi!
>
> I´m using two Opensips as proxys, and they also take decisions about
> redirections of different calls.
> All messages pass through Asterisk.
>
> In a call from 104 to 100, the first thing I do is send the INVITE message
> to Asterisk. Later, after Opensips receives the new INVITE (from Asterisk),
> it decides that this call must be redirected to 200 (which is registered in
> the other proxy). So Opensips (in "opensips.cfg") changes the value of the
> destination (by write $rU="200") and executes the statement: sl_send_reply (
> "302", "Moved Temporarily").
>
> Proxy1 doesn´t receive the new INVITE I suppose it should does. It only
> receives an ACK message from 104 to 100.
>
> Proxy2 receives an INVITE from 200 to 200, but it should be from 104 to 200.
> And it also receives an ACK from 200 to 200.
>
> And the call is made correctly!
>
> I want Asterisk to send the INVITE messages from 104 to 200 to both proxys.
> Any idea?
>
> I hope you understand my problem.....
>
> Thanks!
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro