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 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 > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
