asterisk users - Jan 2005 - Callmanager and Asterisk problem

Hello everybody, i got and asterisk and a CCM configured thru SIP, and in
the sip show peers appear

Name/username    Host            Dyn Nat ACL Mask             Port     Status
CCM              10.60.27.138                255.255.255.255  5060     OK
(1 ms)

but when i enabled sip debug in the CLI got this

------------------------------------------------------------
Sip read:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK784b4a8c
From: "asterisk" <sip:asterisk@10.60.0.136>;tag=as7b541ffe
To: <sip:10.60.27.138>
Call-ID: 315ec3da7feea832650744d256527d5f@10.60.0.136
CSeq: 102 OPTIONS
Content-Length: 0


7 headers, 0 lines
Destroying call '315ec3da7feea832650744d256527d5f@10.60.0.136'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.60.27.138 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK4aaa1423
From: "asterisk" <sip:asterisk@10.60.0.136>;tag=as6f4153c7
To: <sip:10.60.27.138>
Contact: <sip:asterisk@10.60.0.136>
Call-ID: 754ed1fd303984c60ce567ce12b7aa64@10.60.0.136
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 26 Jan 2005 09:15:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 10.60.27.138:5060

--------------------------------------------------------------------

can anybody help me?, what could be the problem?? when i try to call an
ccm extension got the busy signal,

TIA

Edgar

Edgar de Leon wrote:
>Hello everybody, i got and asterisk and a CCM configured thru SIP, and in
>the sip show peers appear
>
>Name/username    Host            Dyn Nat ACL Mask             Port    
Status
>CCM              10.60.27.138                255.255.255.255  5060     OK
>(1 ms)
>  
>
Just a guess, but do you have a username setup in sip.conf? It might be 
that or it might be the "fromuser" setting. Hey, lets see your
sip.conf
please?
-Brett