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

[Gavin Henry]

On 23/05/07, Mojo with Horan & Company, LLC <mojo at horanappraisals.com> wrote:
> Does the non-Asterisk server _answer_ the line? :)

Hi, sorry. I have been away on site doing 8 work ;-)

Yes, it does. We've done a packet trace and it appears that * sends an
ACK back on the wrong port, i.e. not 5605 like a softphone does, in
the SDP session.

>
> 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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