There is a better way. Upgrade to the latest 3.1.3a firmware, which supports PSTN to VoIP gateway calls. Specifically the "Off hook while calling VoIP" option needs to be no.
If you need help configuring it, use the SPA3K w/Asterisk configurator at Voxilla http://voxilla.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 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 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. Take a look at this: http://lists.digium.com/pipermail/asterisk-users/2005-March/097922.html _______________________________________________ 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 _______________________________________________ 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
