I just installed AsteriskNow(Asterisk 1.6.2.11 and FreePBX 2.7.0.6).
I tried setting up my inbound and outbound trunk and they aren’t working. I tried ‘sip show peers’ and their status is ‘Unmonitored’. I added ‘qualify=yes’ and now the statuses say ‘Unknown’. When I try to make a call, I get “==Using SIP RTP TOS bits 184 ==Using SIP RTP CoS mark 5”.
I spoke too soon… I still get “==Using SIP RTP TOS bits 184 ==Using SIP RTP CoS mark 5” when trying to make a call and when receiving a call. The FreePBX statuses page shows my 1 extension online, 1 trunk online, and 0 trunks registered.
Don’t get too focused on those log errors. Get on the console and ‘core set verbose 99’ so that you get every bit of logging out of the dialplan. ‘sip set debug on’ and watch the SIP packets as you attempt to make or receive a call. The only way to fix a problem is to dive in like that.
If the status is showing 0 trunks registered you might want to start by verifying your registration line.
Install the Asterisk Info module; it’s a big help.
Thanks. The only reason I found the log errors relevant was because before, when I made a call to one of my DIDs, nothing would happen. Now that I get an error message, I at least know its reaching my server.
You posted your registration credentials. Might want to change those now.
type=friend in both peer and user settings does not make sense. If you use type=friend, set it up only in the peer section. Otherwise, set up a type=peer and a type=user. But I suspect you don’t have to do that.
Unknown means that it can’t reach the remote server. Better start with firewall settings.