Active SIP dialogs and busy circuits

I have just installed a new PBX system (AsteriskNOW 1.7.1, Asterisk, FreePBX 8 SIP phones are connected and 4 Cisco 186 ATAs. The system seems to be working fine, but I noticed this in the “Asterisk info” -> “Channels” section:

Peer User/ANR Call ID Format Hold Last Message Expiry (None) [email protected] 0x0 (nothing) No Rx: REGISTER (None) [email protected] 0x0 (nothing) No Rx: REGISTER (None) [email protected] 0x0 (nothing) No Rx: REGISTER (None) 5f3a850019f3467 0x0 (nothing) No Init: OPTIONS (None) [email protected] 0x0 (nothing) No Rx: REGISTER (None) [email protected] 0x0 (nothing) No Rx: REGISTER (None) 30[email protected] 0x0 (nothing) No Rx: REGISTER 7 active SIP dialogs

I haven’t seen this on the other systems I have installed. The IP-addresses below to the Cisco ATAs. Is this normal and/or a problem?

One problem I did experience, was that I suddenly couldn’t make outside calls. It had worked fine, but then I started getting an “All circuits are busy” message. All circuits were not busy - up to 30 concurrent calls should be able to pass through the SIP trunk, and I was the only one making calls at the time. I checked the log to make sure the correct number was passed to the trunk (I had been fiddling with dial strings on some phones), and it was. I shut down the system to move it, and when I switched it on again, the problem was gone. I could now dial outside numbers (the same numbers that didn’t work before). What could cause this? And what tests should I make if it happens again?

Best regards,

Mikkel C. Simonsen