[asterisk-users] SIP Dial Command to a non-Asterisk url

[Nick Seraphin]

The 2 most common problems I've seen for "no audio" in one or both
directions is usually either a firewall (which you already said you don't
have) or a CODEC problem.

Make sure both sides are negotiating the same CODEC.  I've often seen
situations where something like the Asterisk server will allow gsm, g711,
etc. and the phone is set for g711, but because gsm was first in the list
on the asterisk side, asterisk was trying to do gsm and the phone wanted
g711 and they wouldn't sync up.  It wasn't until I did a:

	disallow=all
	allow=g711

in sip.conf that it finally started working for me.

That may not be your exact problem, but my guess would be a CODEC issue if
it's not your firewall.

-- Nick


On Wed, 23 May 2007, Gavin Henry wrote:

> Dear All,
> 
> I have a tiny dial plan like:
> 
> [testing]
> exten => 454,s,Ringing()
> exten => 454,n,Wait(4)
> exten => 454,n,Dial(SIP/slee at 192.168.45.183:5605,10)
> exten => 454,n,Hangup
> 
> 
> This connects fine when I dial 454 from any extension in my system,
> but there is never any audio?
> 
> Where can I start to look for debugging this? It's all internal so no
> NAT problems?
> 
> Thanks,
> 
> Gavin.
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