[Asterisk-Users] Sip dosenot fall to default 's' , STRANGE?

[vivek at staff.ownmail.com]

Hello friends, 
 I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I have three SIP phones and one H323 phones connected to asterisk. The problem is that when I dial an invalid extension from H323 phones, I get the invalid extension message with exten => i... in that context but this does not happen with the SIP phones. All I get is something like an engaged tone from the SIP phones. Also I am able to dial and transfer between SIP and H323 phones. I am not able to figure out whats wrong. None of them are behind the NAT. All of them and the asterisk server are on private-ip.
 I also tried  "sip debug" from the command line and dial an invlaid extension from the SIP phone and get nothing but a 
"SIP/2.0 404 Not Found" in the o/p. But it then dosent fall to the exten => i or exten => s.
My conf. files are as under:-

extensions.conf:-
[incoming]
exten => s,1,Answer                                     ; Answer the line.
exten => s,n,Set(TIMEOUT(digit)=5)                      ; Set Digit Timeout to 5 seconds.
exten => s,n,Set(TIMEOUT(response)=10)                  ; Set Response Timeout to 10 seconds.
exten => s,n(restart),BackGround(demo-congrats)         ; Play a congratulatory message.
exten => s,n,WaitExten(5)                               ; Wait for an extension to be dialed.
exten => s,n,Dial(SIP/192.168.1.196,100,t)              , Dial the operator.

exten => i,1,Playback(invalid)                          ; "That's not valid, try again".

[default]
include => incoming                                     ; Instead of demo in the sample, there is incoming.

[testing]
include => parkedcalls

exten => s,1,Playback(invalid)                          ; When this is present, invalid extension from h323 comes here or 
;;; exten => i,1,Playback(invalid)  ;;;even this did not work.           
;;  H323 Phones  ;;
exten => 61,1,Dial(OOH323/192.168.1.194,20|t)      ;ip=h323
;;  SIP Phones   ;;
exten => 62,1,Dial(SIP/62,20|t)                    ;new-gray=sip
exten => 63,1,Dial(SIP/63,20|t)                    ;old-gray=sip
exten => 64,1,Dial(SIP/64,20|t)                    ;ip=sip

ooh323.conf:-
context=testing
disallow=all
allow=ulaw
allow=alaw
dtmfmode=h245alphanumeric
[61]
type=friend
ip=192.168.1.194
context=testing

sip.conf:-
[general]
context=default
bindport=5060 
bindaddr=0.0.0.0
srvlookup=yes
disallow=all                   
allow=alaw
allow=ulaw      
musicclass=default
dtmfmode = rfc2833

[63]
type=friend
context=testing         ; context above where the extensions dialable by this are defined. 
username=63
secret=1234
host=dynamic
defaultip=192.168.1.192 ; ip address of this phone
canreinvite=no
callgroup=1             ; We are in caller groups 1
pickupgroup=1           ; We can do call pick-p for call group 1
;; rest of the sip users are configured in the same way.

Help will be very much appreciated. Kindly help. I am totally confused as to where the fault is. 






With warm regards.

Vivek J. Joshi.

vivek at staff.ownmail.com
Trikon electronics Pvt. Ltd.

--Truth springs from argument amongst friends.