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

[RoLaNd RoLaNd]

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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