[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Raúl Alexis Betancor Santana]

On Saturday 11 July 2009 22:16:27 you wrote:
> Yeah, I suppose so... :)  There is no NAT here, however.  All public IPs.
> The root of the issue isn't NAT fixing, it's that Mediaproxy doesn't seem
> to use properly the new information from a reinvite.

I will not question why are you trying to use Mediaproxy if not for NAT 
fixing .. X-)

> Failing call flow is:
>  PSTN Gateway -> Asterisk w/ reinvites -> Opensips -> SIP Phone*
>
> * Note:  "SIP Phone" is really an Asterisk box with a Polycom behind it,
> but it's not doing anything screwy.  No reinvites from this one.  I can
> reproduce the same behavior with a Sipura or Polycom registered directly to
> Opensips.  It's just much harder to test because I don't have any extra
> public IPs available in my "home" lab.

For properly handling the re-invite, did you call force_rtp_proxy INSIDE the 
in-dialog procces ?

-- 
Raúl Alexis Betancor Santana
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