asterisk users - Feb 2011 - Asterisk 1.8.3-rc3 and one way audio

I've just installed 1.8.3-rc3 on a test server as we really needed that
deadlock involving REFER fix on our server but now I'm having an odd
issue with one way audio with a specific type of call.

If I do extension to extension calls there is full 2 way audio.

If I route in an incoming call through numbers provided by our SIP
provider there is no inbound audio (mobile to * SIP extension) but there
is outbound audio (* SIP extension to mobile).

If I route a call through our production server (1.4.17 debian) to a
second identity on the same SIP phone as the previous condition there is
perfect 2 way audio.

I did suspect it might have been the firewall on the test server but I
did the same call with the firewall turned off and still only had one
way audio.

Has anyone else experienced anything like this?
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

On Mon, 2011-02-28 at 13:40 +0000, Ishfaq Malik wrote:
> I've just installed 1.8.3-rc3 on a test server as we really needed that
> deadlock involving REFER fix on our server but now I'm having an odd
> issue with one way audio with a specific type of call.
> 
> If I do extension to extension calls there is full 2 way audio.
> 
> If I route in an incoming call through numbers provided by our SIP
> provider there is no inbound audio (mobile to * SIP extension) but there
> is outbound audio (* SIP extension to mobile).
> 
> If I route a call through our production server (1.4.17 debian) to a
> second identity on the same SIP phone as the previous condition there is
> perfect 2 way audio.
> 
> I did suspect it might have been the firewall on the test server but I
> did the same call with the firewall turned off and still only had one
> way audio.
> 
> Has anyone else experienced anything like this?
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

Ish

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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?
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