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 <[email protected]> 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:0123400...@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 > [email protected] > > _______________________________________________ > -- 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 _______________________________________________ -- 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
