[Asterisk-Users] What might be blocking RTP

[Howard Lowndes]

On Sun, 2004-12-12 at 03:46, Eric Wieling aka ManxPower wrote:
> Howard Lowndes wrote:
> > When I make a call from a SIP phone to a speaking extension on *, such
> > as one that speaks digits or similar, when I monitor * in very verbose
> > mode I can see it running through the routine associated with the
> > extension, but I am getting no RTP data stream back to the phone.
> > 
> > Does the machine housing * need a sound card?
> > Does it need OSS or ALSA modules installed?
> > What actually generates the RTP data stream?
> > 
> 
> You don't need a soundcard.

That's what I thought.
> 
> Is Asterisk behind NAT?

No, this is a local network.

>   If so look at localnet= and externip= in 
> sip.conf and look into portforwarding and rtp.conf.

It won't need portforwarding being a local network.
I might just check out rtp.conf.

>   Remember AUDIO on 
> SIP/H323/MGCP/SCCP are sent using the RTP protocol.

Yes, I am aware of that, and that is what I am not getting back from *.

>   SIP is just a 
> signaling protocol.

...aware of that too.

-- 
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
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