vivek@staff.ownmail.com
2005-Nov-24 04:35 UTC
[Asterisk-Users] Sip dosenot fall to default 's' , STRANGE?
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@staff.ownmail.com Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends.