it took about one week to get asterisk 1.2 with Freepbx 2.2.1 running. Now i have a problem wich i cannot solve by myself. I always tried many various combinations of trunkdefinitions and inbound and outbound routes.
Incoming Sip Calls are showed on my console but not extension is ringing. here is the output of such a try:
Retransmitting #5 (no NAT) to 184.108.40.206:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 220.127.116.11;branch=z9hG4bK2fa84ca5
From: 01726435398 sip:firstname.lastname@example.org;tag=as5922fa80
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1ac95f79"
Is there any errormessage or can somebody see that anything is going wrong?
Please help me to get rid of the Problem.
The Machine is running on a Centos vServer from my provider without nat.