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

[Marc Leurent]

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
> >>>>
> >>>> _______________________________________________
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> >>>>
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> >>
> > 
> > 
> > 
> 
> 



-- 
-- --
Marc LEURENT
lftsy at leurent.eu