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

[Mojo with Horan & Company, LLC]

Does the non-Asterisk server _answer_ the line? :)

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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