[Asterisk-Users] Re: Connecting Sip phone to asterisk.

[joosfamily at speakeasy.net]

The phone is configured as:
IP Phone Number: 1201
Username: 1201
Password: <password>
Service Address: 192.168.0.104

Sip.conf is configured as:
[1201]
type=friend
username=1201
secret=<password>
mailbox=1201
host=192.168.0.99


To keep the redundant data down, here is what the sip debug shows:

Retransmitting #5 (no NAT):
INVITE sip:1201 at 192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077
From: "1202" <sip:1202 at 192.168.0.104>;tag=as166994fa
To: <sip:1201 at 192.168.0.99>
Contact: <sip:1202 at 192.168.0.104>
Call-ID: 5ac70b3e78833fd40669cb867ba3ccb7 at 192.168.0.104
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 08 Jan 2005 20:22:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6552 6552 IN IP4 192.168.0.104
s=session
c=IN IP4 192.168.0.104
t=0 0
m=audio 11670 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.0.99:5060
Jan  8 12:22:25 WARNING[6552]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 5ac70b3e78833fd40669cb867ba3ccb7 at 192.168.0.104 for seqno 102 (Critical Request)
Destroying call '5ac70b3e78833fd40669cb867ba3ccb7 at 192.168.0.104'