[Asterisk-Users] Sipura 3000 help

Adrian A adrianvoip at gmail.com
Thu Jun 16 15:00:30 MST 2005


Right, turns out I am an idiot and I do have Asterisk running on 5070
instead of 5061.  It's all working.
Now, if I could find out why calls coming from PSTN have horrible
voice quality....

On 6/16/05, Luki <lugosoft at gmail.com> wrote:
> > I can see on tcpdump traces that the Invite packets
> > do go to through to the asterisk machine on port 5061,
> > but it's not picking them up.  sip debug does not show
> > any packets either.
> 
> That would imply that the Sipura config is fine, but your Asterisk
> setup is not listening at the right interface, IP or port. If the
> extension was invalid or there was an authentication issue, Asterisk
> should send a reply message not just ignore it...
> 
> --Luki
>



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