I’m running 2.10 of the distro behind a sonicwall TZ-210. I’ve hacked on the firewall so much to get various things working properly that this weekend I finally reset it to factory defaults and rebuilt my configs from scratch to make things alot cleaner, and so I’d force myself to make some notes.
I have everything working as before, but find one thing I’m trying to accomplish a bit confusing.
I Have 2 Wan Connections, 2 ISP’s
I have sip Trunks from Vitelity, Routing to WAN 1, if unavailible to WAN 2
If WAN 1 is up all is well.
When WAN 1 Goes Down, Vitelity Routes Calls to WAN 2, PBX answers, I have audio from PBX to PSTN but not from PSTN to PBX. Call Disconnects after 30 seconds Due to lack of RTP Traffic
NOTICE[4140]: chan_sip.c:25002 check_rtp_timeout: Disconnecting call ‘SIP/vitel-inbound-00000479’ for lack of RTP activity in 31 seconds
I can Make calls Out from PBX but no audio either way.
NOW… I’m pretty sure this is a NAT issue, Because in Asterisk Sip Settings I have the Static IP as my WAN 1 IP.
My first question, can you have two static IP’s associated?
I assume selecting dynamic and using a DDNS provider and adjusting the timeout is 1 way to resolve it the issue. I have sysadmin Pro, and see the two entries Smart DDNS Address and External DDNS Name… is one of these what I should put as Dynamic Host in Asterisk SIP Settings, or signup for another DDNS service?
In ASterisk Sip Settings, the refresh rate can be keyed in, 30 seconds would be my preference… But in Sysadmin>DDNS Update intervals are 5min, 15min, and 1hr. When I set to 5min it reverts to 15 min Everytime. 5 Minutes is too long for a Failover… am I missing something here?