[Asterisk-Users] Sip Phone Won't Login...

[joosfamily at speakeasy.net]

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 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 192.168.0.101>;tag=as167cf09c
To: <sip:Dan1 at 192.168.0.101>
Call-ID: 6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:1201 at 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 at 127.0.0.1 for seqno 115 (Critical Request)
Destroying call '6df3875f689537fb23ca5b6d0e6a28f1 at 127.0.0.1'