Tuesday, September 08, 2009

SIP trunks- Asterisk to Future-nine

Dude...if your inbound calls from your PSTN SIP provider are hit-or-miss, might want to try adding qualify=yes to your Asterisk SIP config for the trunk. Mine are solid now. My firewall must have been closing up the NAT association. My stupid firewall is not very conifgurable, so I can't set the timeout there.

With qualify=yes, traffic is ping-ponged every 60 seconds to keep the firewall happy during idle times.

1 comment:

Future Nine VoIP said...

Good point. I added the following comment to the SIP INFO page:

; set qualify=yes if you have an incoming number and are behind a NAT!

I ran into your blog by accident, but thanks for the reminder. :)

Nitzan Kon, CEO
Future Nine Corporation