Thank you, this is exactly what I needed!! In order to Dial any number to a registered peer, I just have to enter Dial(SIP/anynum...@sippeername) Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit : > The Request URI generated in an INVITE originated by Asterisk is > governed entirely by the parameters passed to Dial(). > > For example: > > Dial(SIP/1...@peer_name) > > ... will generate a Request URI of > 1...@host.or.ip.of.sip.conf.peer.named.peer_name. > > It is also possible to send requests to hosts that are not explicitly > defined in sip.conf, with the caveat that only background [general] > sip.conf settings will then apply: > > Dial(SIP/1...@ip.of.peer.not.in.sip.conf) > > Marc Leurent wrote: > > > Hello, > > it is not an OpenSIPs problem I have, it's an Asterisk one, > > I would like to change the URI in message generated by Asterisk. > > Thanks > > > > Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit : > >> Modify the $ru pseudovariable or use rewritehostport() out of core. > >> > >> This is not the right mailing list. This belongs on the > >> OpenSIPS/OpenSER lists. > >> > >> There is also a mailing list we operate called SER-Asterisk-Interwork > >> that is specifically intended to address SER* / Asterisk integration > >> issues: > >> > >> http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork > >> > >> * Anything from the [Open]SER family. > >> > >> lftsy wrote: > >> > >>> 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 <lf...@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: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 > >>>> lf...@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 > >>> _______________________________________________ > >>> -- 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 > >> > > > > > > > > -- -- -- Marc LEURENT lf...@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
