Asterisk help

[firestorm_v1]

context=custom-voipo

Dswartz:
I tried your config but I have never seen this context used. How did you define it?

scott2020:
I tried your config too and I don't know if I've taken a step forward or a step back. The line rings once, and then I get a male voice that says "The number or code that you have dialed is incorrect. Please check the number or code and try again."

I noticed that the log goes nuts when I call it, and it looks like the call IS hitting the server:
[Jun 17 16:10:18] --- (20 headers 16 lines) ---
[Jun 17 16:10:18] Sending to 67.228.251.106 : 5060 (no NAT)
[Jun 17 16:10:18] Using INVITE request as basis request - 420301830_32241846@64.156.174.74
[Jun 17 16:10:18] Found peer 'VOIPo'
[Jun 17 16:10:18] Found RTP audio format 0
[Jun 17 16:10:18] Found RTP audio format 18
[Jun 17 16:10:18] Found RTP audio format 4
[Jun 17 16:10:18] Found RTP audio format 101
[Jun 17 16:10:18] Peer audio RTP is at port 67.228.251.106:43066
[Jun 17 16:10:18] Found audio description format PCMU for ID 0
[Jun 17 16:10:18] Found audio description format G729 for ID 18
[Jun 17 16:10:18] Found audio description format G723 for ID 4
[Jun 17 16:10:18] Found audio description format telephone-event for ID 101
[Jun 17 16:10:18] Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Jun 17 16:10:18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jun 17 16:10:18] Peer audio RTP is at port 67.228.251.106:43066
[Jun 17 16:10:18] Looking for {My Voipo #} from-sip (domain 192.168.0.14)
[Jun 17 16:10:18]


Another question is that I have the catchall route set to Extension 200 (softphone). Should I set this to Ring Group 600 (Ring all phones) or should I wait until I can confirm its operation first?

I can't thank you all enough for your help with this. I am very excited at the possibility of getting this up and running.