[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Jeff Pyle]

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.

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.


- Jeff



On 7/11/09 4:52 PM, "Raúl Alexis Betancor Santana"
<rabs at dimension-virtual.com> wrote:

> If you want to improve readability .. just don't use IP's from a same range in
> a capture that is supposed to be about NAT fixing  ... ;-)
> 
> After a first read ... your call flow is Asterisk -> OpenSIPS -> Asterisk ->
> SIP Phone ? ... or Asterisk -> OpenSIPS -> SIP Phone (beging the same NAT
> router as Asterisk) ?
> 
>