[asterisk-users] Asterisk 1.8.3-rc3 & 1.8.3 and one way audio

[Ishfaq Malik]

On Tue, 2011-03-01 at 10:08 -0600, Terry Wilson wrote:
> 
> On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote:
> > 
> > Seeing that 1.8.3 had been released I updated our main test server
> > to
> > that from 1.8.2.2 using the digium yum repo.
> > 
> > All audio had been working fine on this server before the update but
> > after the update I experienced the same as I did with rc3.
> > 
> > Internal ext to ext calls are fine. 
> > 
> > Outbound calls to mobile networks via our SIP provider are fine.
> > 
> > Inbound calls via our SIP provider have one way audio.
> > 
> > The servers are CentOs 5.5 and we are using RealTime architecture.
> > 
> > Any thoughts would be appreciated
> > 
> 
> One-way audio is almost always NAT related. Does the Asterisk server
> have a public IP?

I think this is NAT related. The server has it's own public IP and is in
a data centre. The extension is in out office and behind a NAT.
If I do a sip show peer then the Addr->IP shows our office real world IP
address but the reg contact shows my phones internal network IP Address.
The really strange thing is that there is no section for NAT which I
would expect to show as always.

The sip peer data is stored in a MySQL table with the column name nat
and a value of yes. I so far have not been able to find a definitive
table definition for asterisk 1.8 and have been using the one for 1.6

Does anyone know if the nat column has been changed in the sip realtime
table?

I'd even happily have a rummage in the source code if anyone would point
me in the direction of the right files to look at.

Thanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062