[Asterisk-Users] Problem with extensions and when channel is unavailable

[Jean-Yves Avenard]

Hello

I used to have an extension like this which worked fine with asterisk  
1.0.7
I first dial to see if an IAX phone is present, if not I would try on  
SIP instead

exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout
exten=s,2,Goto(s-${DIALSTATUS},1)

; Default action
exten=s,200,DBget(temp=CFBS/${ARG1})  ; Get CFBS key, if not  
existing, goto 301
exten=s,201,Dial(Local/${temp}@sip-transfer/n) ; Forward on unavailable
exten=s,202,VoiceMail,u${ARG2}
exten=s,203,hangup()
exten=s,301,Goto(202)   ; entry doesn't exist -> Voicemail

exten=s-CHANUNAVAIL,1,Goto(s,400)
exten=s-CONGESTION,1,Goto(s,200)
exten=s-NOANSWER,1,Goto(s,200)
exten=s-.,1,Goto(s,200)

;Now try SIP
exten=s,400,Dial(SIP/ipp${ARG3},20,tr) ; 20sec timeout
exten=s,401,Goto(s2-${DIALSTATUS},1)
exten=s2-CHANUNAVAIL,1,Goto(s,200)
exten=s2-CONGESTION,1,Goto(s,200)
exten=s2-NOANSWER,1,Goto(s,200)
exten=s2-.,1,Goto(s,200)


If the IAX channel didn't exist or wasn't connected, it would jump to  
s-CHANUNAVAIL which jumps to priority 400 and try SIP

In CVS-HEAD it now goes to s-CONGESTION which is definitely not what  
I want as it goes in this case to voicemail

s-DIALSTATUS has changed now? is there a documentation on how it  
should work now?

Jean-Yves

---
Jean-Yves Avenard
Hydrix Pty Ltd - Embedding the net
www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct  
+61 3 8573 5200

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