asterisk users - Feb 2005 - why asterisk is replying 404 Not Found

[3000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
                                                      
                                          
[2000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw

i have declared these two users 3000 and 2000. they
are registering successfully.

problem is that when i am clling for 3000 to 2000. it
is replying me 404 Not found and printing (found user
'3000' looking for 2000 in default). and when i am
calling from 2000 to 3000. it is replying me 404 not
found and printing (found user '2000' looking for 3000
in default).

can any one tell me what is the problem. i dont know
what is problem. i have just compilied my
asterisk(stable) using LAN.

thanks

Found user '3000'
Looking for 2000 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.109;branch=z9hG4bKc0a8006d0131c9b14209a71a2c53e29900000004
From:
"root"<sip:3000@192.168.0.203>;tag=4122263641472432750
To: <sip:2000@192.168.0.203>;tag=as2f7c300d
Call-ID:
F81AB0F0-1DD1-11B2-91B7-F323B891C30C@192.168.0.109
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@192.168.0.203>
Content-Length: 0


	
		
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What does your extensions.conf file look like?
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf

Dan

Kamran Ahmad wrote:
>[3000]
>type=friend
>dtmfmode=INFO
>insecure=yes
>canreinvite=no
>auth=plaintext
>host=dynamic
>allow=ulaw
>                                                      
>                                          
>[2000]
>type=friend
>dtmfmode=INFO
>insecure=yes
>canreinvite=no
>auth=plaintext
>host=dynamic
>allow=ulaw
>
>  
>
>i have declared these two users 3000 and 2000. they
>are registering successfully.
>
>problem is that when i am clling for 3000 to 2000. it
>is replying me 404 Not found and printing (found user
>'3000' looking for 2000 in default). and when i am
>calling from 2000 to 3000. it is replying me 404 not
>found and printing (found user '2000' looking for 3000
>in default).
>
>  
>
>can any one tell me what is the problem. i dont know
>what is problem. i have just compilied my
>asterisk(stable) using LAN.
>
>thanks
>
>Found user '3000'
>Looking for 2000 in default
>Reliably Transmitting (no NAT):
>SIP/2.0 404 Not Found
>Via: SIP/2.0/UDP
>192.168.0.109;branch=z9hG4bKc0a8006d0131c9b14209a71a2c53e29900000004
>From:
>"root"<sip:3000@192.168.0.203>;tag=4122263641472432750
>To: <sip:2000@192.168.0.203>;tag=as2f7c300d
>Call-ID:
>F81AB0F0-1DD1-11B2-91B7-F323B891C30C@192.168.0.109
>CSeq: 1 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:2000@192.168.0.203>
>Content-Length: 0
>
>
>	
>		
>__________________________________ 
>Do you Yahoo!? 
>Yahoo! Mail - You care about security. So do we. 
>http://promotions.yahoo.com/new_mail
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  
>

> [3000]
> type=friend
> dtmfmode=INFO
> insecure=yes
> canreinvite=no
> auth=plaintext
> host=dynamic
> allow=ulaw
>                                                       
>                                           
> [2000]
> type=friend
> dtmfmode=INFO
> insecure=yes
> canreinvite=no
> auth=plaintext
> host=dynamic
> allow=ulaw
> 
> i have declared these two users 3000 and 2000. they
> are registering successfully.
> 
> problem is that when i am clling for 3000 to 2000. it
> is replying me 404 Not found and printing (found user
> '3000' looking for 2000 in default). and when i am
> calling from 2000 to 3000. it is replying me 404 not
> found and printing (found user '2000' looking for 3000
> in default).
It may help you a lot to add "context=from-sip" in each of the
two above. Then in your extensions.conf define this context with
the appropriate "exten => ..." entries for your dialplan.

It would appear you are getting caught with not understanding the
use of contexts, and your existing sip.conf entries are defaulting
to the "default" context and you don't see/know that.

So, given the above, your extensions.conf would have something
like this in in:

[from-sip]
exten => 3000,1,Dial(SIP/3000,15,r)
exten => 3000,2,Voicemail(u3000)
exten => 3000,102,Voicemail(b3000)
exten => 3000,103,Hangup

exten => 2000,1,Dial(SIP/2000,15,r)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup

Hope that helps...

[default]
;
; By default we include the demo.  In a production
system, you 
; probably don't want to have the demo there.
;
include => demo
exten => 3000,1,Dial(SIP/${EXTEN})
exten => 2000,1,Dial(SIP/${EXTEN})



		
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thanks

it is register and receiving the invite. some time my
user agent (i am sjphone) is sending invalid address
in his contact and SDP. then i try to call form
another ua it i transmitting invite to invalid
address.




		
__________________________________ 
Do you Yahoo!? 
The all-new My Yahoo! - What will yours do?
http://my.yahoo.com