Hey Peoples, I just got my paws on a KE1020A Phone and all it is doing when I plug it in is: 1201 Wait Login... Sip.conf [1201] type=friend username=1201 secret=<password> host=216.254.10.183 mailbox=1201 context=intern canreinvite=yes dtmfmode=rfc2833 nat=1 register => 1201:<password>@216.254.10.183/1201 One side note, The KE1020A does not have NAT capabilities, but I am running NAT behind my firewall. Below is the sip debug information. I hope this helps! localhost*CLI> sip debug ip 192.168.0.101 SIP Debugging Enabled for IP: 192.168.0.101 Jan 6 23:10:18 NOTICE[6859]: chan_sip.c:4059 sip_reg_timeout: Registration for 'Dan1@192.168.0.101' timed out, trying again 11 headers, 0 lines Reliably Transmitting: REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 (no NAT) to 192.168.0.101:5060 Retransmitting #1 (no NAT): REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 to 192.168.0.101:5060 Retransmitting #2 (no NAT): REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 to 192.168.0.101:5060 Retransmitting #3 (no NAT): REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 to 192.168.0.101:5060 Retransmitting #4 (no NAT): REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 to 192.168.0.101:5060 Retransmitting #5 (no NAT): REGISTER sip:192.168.0.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK4c154209 From: <sip:Dan1@192.168.0.101>;tag=as167cf09c To: <sip:Dan1@192.168.0.101> Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 CSeq: 115 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:1201@192.168.0.104> Event: registration Content-Length: 0 to 192.168.0.101:5060 Jan 6 23:10:24 WARNING[6859]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1 for seqno 115 (Critical Request) Destroying call '6df3875f689537fb23ca5b6d0e6a28f1@127.0.0.1'