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

[RoLaNd RoLaNd]

hey i just added it in ocntext spa, now it picks upt he phone, but when i try accessing extension 110 it gives me tht last error and then the line hangs up after a few seconds..
 
  == Connect attempt from '127.0.0.1' unable to authenticate    -- Executing [100 at spa:1] Goto("SIP/101-b5f65f20", "sipura-line|100|1") in new stack    -- Goto (sipura-line,100,1)    -- Executing [100 at sipura-line:1] Answer("SIP/101-b5f65f20", "") in new stack    -- Executing [100 at sipura-line:2] WaitExten("SIP/101-b5f65f20", "10") in new stack[May 26 19:53:26] WARNING[27678]: pbx.c:2494 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context 'sipura-line'


From: roberto.milani at sbcglobal.netTo: asterisk-users at lists.digium.comDate: Mon, 26 May 2008 09:23:54 -0700Subject: Re: [asterisk-users] sipura 3102 dial plan : (S0<:100>) not being answered on asterisk!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=1234allow=allhost=dynamictype=friendcontext=sipura-line[101]secret=1234allow=allhost=dynamictype=friendcontext=spa[103]secret=1234allow=allhost=dynamictype=friendcontext=spaextensions.conf:[sipura-line]exten => 100,1,Answer() ; Answer inbound callsexten => 100,2,Playback(silence/1)exten => 100,3,Background(/etc/asterisk/simzy.wav) ; input an extensionexten => 100,n,WaitExten(5) ; Adjust wait, default 5 secexten => 100,n,Goto(internal,${EXTEN},1) ; Goto the correct extensionexten => 100,n,Hangup() ; End the call[spa]Exten => _1XX,1,Dial(SIP/${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 ringingand 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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