I have spoken to quickly, Usually Asterisk on an incoming call sends an INVITE "Reg.Contact Number"@"Reg Contact IP" to the Peer IP. With the command you gave me, it is possible to send an INVITE "othernumber"@"Peer IP" to the Peer IP. What I would like to do is to send INVITE "othernumber"@"Reg Contact IP" to the Peer IP in order for the request to be forwarded by the proxy!
Is it possible to do something like: Dial(SIP/"<sip:1...@192.168.10.125:5060>"@1003 ) in Order to send INVITE "1...@1005 IP" to 1003 device IP Thanks! Le Monday 23 March 2009 12.03:55 Marc Leurent, vous avez écrit : > 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
