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

[Alex Balashov]

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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>>
> 
> 
> 


-- 
Alex Balashov
Evariste Systems
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