[asterisk-users] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk

[lftsy]

Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent by
Asterisk (first line of SIP packets)

Thanks for your time!
++


On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent <lftsy at leurent.eu> wrote:
> Hello All,
> I have a little complicated question about the Dial command.
> I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered
> on Asterisk servers.
> Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs
> server. Everything works except for trunk numbers:
> 
> For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg.
> Contact" is the IP where the proxy will relay the packet to reach the
UAC.
> 
> Ex: with a trunk 0123400010 -> 0123400019 with 0123400010 as the sip
peer.
> When a number from a trunk is called, like 0123400019  the "Reg. Contact"
> of the main number is not used.
> 
> For the time being, I use Dial(SIP/0123400010/0123400019) but it It sends
> an
> INVITE sip:0123400019 at proxyIP to the proxy
> 
> whereas it should send
> INVITE sip:0123400019@"Reg. Contact of the main number" to the proxy
> 
> So I'm trying use the Dial Command with
> Dial(SIP/0123400010/0123400019@"Reg. Contact of the main number") but it
> doesn't work
> 
> Have you got any idea how to rewrite the IP of the URI sent?
> Thanks!
> 
> --
> -- --
> Marc LEURENT
> lftsy at leurent.eu
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users