[Asterisk-Users] help on 2 X-Lite: call failed: 404 not found

[lee tance]

Dear All, 

     I installed an Asterisk on a linux PC, and X-Lite on two Windows
PCs, all in a LAN.

     But, when I make phone call from one X-Lite to another, I always get 
            
                   Call Failed: 404 not found. 


      Here is my sip.conf:
 
                  [Phone1]
                  type=friend
                  host=dynamic
                  ;defaultip=192.168.1.103
                  dtmfmode=rfc2833
                  context=SIP
                  callerid = "Me" <2124>

                  [Phone2]
                  type=friend
                  host=dynamic
                  ;defaultip=192.168.1.101
                  dtmfmode=rfc2833
                  context=SIP
                  callerid = "Mini Me" <2123>

             Following is my extensions.conf:
                  exten => 2124,1,Dial(SIP/Phone1,20,tr)
                  exten => 2123,1,Dial(SIP/Phone2,20,tr)

             Here is the Asterisk Sip debug info:

                              <-- SIP read from 192.168.2.103:5060:
INVITE sip:2123 at 192.168.2.120 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.103:5060;rport;branch=z9hG4bK5C01A7C11D6711DA92170800460D92CD
From: 1 <sip:Phone1 at 192.168.2.120>;tag=570805602
To: <sip:2123 at 192.168.2.120>
Contact: <sip:Phone1 at 192.168.2.103:5060>
Call-ID: 5C01A7C0-1D67-11DA-9217-0800460D92CD at 192.168.2.103
CSeq: 24637 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 297

v=0
o=Phone1 22215362 22215384 IN IP4 192.168.2.103
s=X-Lite
c=IN IP4 192.168.2.103
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (11 headers 13 lines)---
Using INVITE request as basis request -
5C01A7C0-1D67-11DA-9217-0800460D92CD at 192.168.2.103
Sending to 192.168.2.103 : 5060 (non-NAT)
Found user 'Phone1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.103:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined
- 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 2123 in SIP
Sep  4 23:21:51 NOTICE[4337]: pbx.c:1680 pbx_extension_helper: Cannot
find extension context 'SIP'
Reliably Transmitting (no NAT) to 192.168.2.103:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.2.103:5060;branch=z9hG4bK5C01A7C11D6711DA92170800460D92CD
From: 1 <sip:Phone1 at 192.168.2.120>;tag=570805602
To: <sip:2123 at 192.168.2.120>;tag=as26bf2947
Call-ID: 5C01A7C0-1D67-11DA-9217-0800460D92CD at 192.168.2.103
CSeq: 24637 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2123 at 192.168.2.120>
Content-Length: 0


---

<-- SIP read from 192.168.2.103:5060:
ACK sip:2123 at 192.168.2.120 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.103:5060;rport;branch=z9hG4bK5C01A7C11D6711DA92170800460D92CD
From: 1 <sip:Phone1 at 192.168.2.120>;tag=570805602
To: <sip:2123 at 192.168.2.120>;tag=as26bf2947
Contact: <sip:Phone1 at 192.168.2.103:5060>
Call-ID: 5C01A7C0-1D67-11DA-9217-0800460D92CD at 192.168.2.103
CSeq: 24637 ACK
Max-Forwards: 70
Content-Length: 0


        

                 Could you help to find out what's my problem?

                 Thanks a lot!


Tance