[asterisk-users] sipura 3102 dial plan : (S0<:100>) not being answered on asterisk!

[Roberto Milani]

Why do you expect anything different?
> Exten => _1XX,1,Dial(SIP/${EXTEN})

Dial means Dial
and it is Dialing.

> Exten => _1XX,1,Dial(SIP/${EXTEN})

in spa context might be changed into

exten => _1XX,1,GoTo(sipura-line,${EXTEN},1)

Ciao
Roberto

On May 26, 2008, at 8:34 AM, RoLaNd RoLaNd wrote:

> sip.conf:
>
>
> [100]
>
> secret=1234
> allow=all
> host=dynamic
> type=friend
> context=sipura-line
>
> [101]
>
> secret=1234
> allow=all
> host=dynamic
> type=friend
> context=spa
>
> [103]
>
> secret=1234
> allow=all
> host=dynamic
> type=friend
> context=spa
>
>
>
>
>
> extensions.conf:
>
>
> [sipura-line]
> exten => 100,1,Answer() ; Answer inbound calls
> exten => 100,2,Playback(silence/1)
> exten => 100,3,Background(/etc/asterisk/simzy.wav) ; input an  
> extension
> exten => 100,n,WaitExten(5) ; Adjust wait, default 5 sec
> exten => 100,n,Goto(internal,${EXTEN},1) ; Goto the correct extension
> exten => 100,n,Hangup() ; End the call
>
> [spa]
> Exten => _1XX,1,Dial(SIP/${EXTEN})
> exten => _0.,1,Dial(SIP/101/${EXTEN:1})
>
>
>   sip debug:
>
>
> == Connect attempt from '127.0.0.1' unable to authenticate
>     -- Executing [100 at spa:1] Dial("SIP/101-b5f65a78", "SIP/100") in  
> new stack
>     -- Called 100p*CLI>
>     -- SIP/100-08234f40 is ringing
>
>
>
> and on sipura 3102, ive set in the dial plan that PSTN to VOIP is  
> directed to: (S0<:100>)
>
>
> the phone keeps on ringing, on 100 i could see it ringing in the CLI  
> as u see above, but it doesnt get picked up to run the audio msg ive  
> set in the extensions.conf
>
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