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 >> > > > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 _______________________________________________ -- 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