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

[Marc Leurent]

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:1005 at 192.168.10.125:5060>"@1003 )
in Order to send INVITE "1005 at 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/ANYNUMBER at 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/1234 at peer_name)
> > 
> > ... will generate a Request URI of 
> > 1234 at 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/1234 at 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 <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
> > >>> _______________________________________________
> > >>> -- 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
lftsy at leurent.eu