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 <[EMAIL PROTECTED]> wrote:
> > I can see on tcpdump traces that the Invite packets
> > do g
oxilla.com/spa3kasterisk.php is adequate.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Thursday, June 16, 2005 4:42 PM
To: Adrian A; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura 3000 help
> Anyone kno
> Anyone know what I need to do to get the FXO port on the SPA 3000 to
> forward calls to Asterisk? My Asterisk is running on port 5061 and I
> set the dial plan on the device to forward to [EMAIL PROTECTED]:5061 but
> Asterisk is not picking it up. I can see on tcpdump traces that the
> Invite
> 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 inter