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

[Gavin Henry]

On 23/05/07, Nick Seraphin <asterisk at eaglequest.com> wrote:
>
>
> 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.

I'll check this out, thanks.

>
> -- 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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