asterisk users - Jan 2005 - Connecting Sip phone to asterisk.

I am having a major dillema here, I have been trying to get my sip phone (hard
phone) to communicate with the asterisk server. Below is my configuration:

sip.conf

[1201]
type=friend
username=1201
secret=<password>
mailbox=1201
host=dynamic

[1202]
type=friend
username=1202
secret=<password>
mailbox=1202
host=dynamic

extensions.conf
exten => 1201,1,Dial(SIP/1201,9,rt)
exten => 1201,n,Voicemail(u1201)
exten => 1201,s+1,Hangup

exten => 1202,1,Dial(SIP/1202,9,rt)
exten => 1202,n,Voicemail(u1202)
exten => 1202,s+1,Hangup

Anyone have any ideas on if there is something that is missing? Also, the model
of the phone is KE1020A

Try debugging sip with "debug sip" at the CLI.  This will provide you 
with more detail on what is going wrong.

Scott

joosfamily@speakeasy.net wrote:
>I am having a major dillema here, I have been trying to get my sip phone
(hard phone) to communicate with the asterisk server. Below is my configuration:
>
>sip.conf
>
>[1201]
>type=friend
>username=1201
>secret=<password>
>mailbox=1201
>host=dynamic
>
>[1202]
>type=friend
>username=1202
>secret=<password>
>mailbox=1202
>host=dynamic
>
>extensions.conf
>exten => 1201,1,Dial(SIP/1201,9,rt)
>exten => 1201,n,Voicemail(u1201)
>exten => 1201,s+1,Hangup
>
>exten => 1202,1,Dial(SIP/1202,9,rt)
>exten => 1202,n,Voicemail(u1202)
>exten => 1202,s+1,Hangup
>
>Anyone have any ideas on if there is something that is missing? Also, the
model of the phone is KE1020A
>
>
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-- 
Scott Henderson
===========================================================================Finite
Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
============================================================================

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@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077
From: "1202" <sip:1202@192.168.0.104>;tag=as166994fa
To: <sip:1201@192.168.0.99>
Contact: <sip:1202@192.168.0.104>
Call-ID: 5ac70b3e78833fd40669cb867ba3ccb7@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@192.168.0.104 for seqno 102
(Critical Request)
Destroying call '5ac70b3e78833fd40669cb867ba3ccb7@192.168.0.104'

Here is the discovery that I have made.

* I can Ping my phone w/o any problems.
* Near as I can tell, I have set up the phone correctly in sip.conf
* When I set up sip debug and/or sip debug ip and I reboot my phone, I 
do not see any activity from the server from my phone.

Username     Secret           Accountcode     Def.Context     ACL  NAT
1202            <password>                           default             
No     No
1201            <password>                           default             
No     No

Name/username    Host               Dyn    Nat ACL Mask             
Port      Status
1202/1202           (Unspecified)                   
0.0.0.0                    5060     Unmonitored
1201/1201           192.168.0.10                  255.255.255.255    
5060     Unmonitored