Hello,

Thanks for the advice, although I don't think it will help my issue. The 200
OK and other packets are returned to the OpenSIPS server perfectly fine.

The problem is that the destination phone is showing the call with the
Asterisk IP address in it's history, and so if the user chooses to place a
return call to that address (eg returning a missed call) it goes directly to
Asterisk, bypassing OpenSIPS.
I'm not actually sure if the phone is displaying the From address or the
Contact address so it would be useful to make both mention the OpenSIPS IP.

Thanks,

On 31 March 2011 18:41, federico cabiddu <[email protected]> wrote:

> Hi,
> I guess that what you want is that the endpoint receiving the INVITE
> sends the 200 OK reply and the subsequent in-dialog messages to
> OpenSIPs
> For the first point you need to do nothing as the reply should be sent
> to the address in the topmost Via header (in this case OpenSIPs'
> address). For the second, call record_route() in OpenSIPs script
> before sending out the INVITE. The endpoint will then send in-dialog
> messages (BYE, re-INVITE) to OpenSIPs.
> Hope thsi can help you.
>
> Regards,
>
> Federico
>
> 2011/3/31 David Cunningham <[email protected]>:
> > Hi all,
> >
> > We have an Asterisk server sending INVITEs via OpenSIPS with a "Contact:
> > <sip:[email protected]>" header where 123.456.789.012 is the Asterisk
> > server IP. I want the INVITE's Contact as it leaves OpenSIPS to say the
> > OpenSIPS IP address instead.
> >
> > How can I do this? From my understanding of the docs, this is the
> opposite
> > of fix_nated_contact.
> >
> > Thanks for any advice.
> >
> > --
> > David Cunningham, Voisonics
> > http://voisonics.com/
> > US toll-free: +1 888 842 2720
> > UK: +44 (0) 20 3298 1642
> > Australia: +61 (0) 2 8063 9019
> >
> >
> > _______________________________________________
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> >
> >
>
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-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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