> Extensions.conf
> [mainmenu]
> exten => 501,1,Answer
> exten => 501,n,Wait(2)
> exten => 501,n,Playback(velkommen_abacus)
> exten => 501,n,Set(Loop=0)
> exten => 501,n,While($[${Loop} < 3])
> exten => 501,n,Background(tast123vent_)
> exten => 501,n,WaitExten(5)
> exten => 501,n,Set(Loop=$[${Loop}+1])
> exten => 501,n,(LoopEnd),EndWhile

This should be:
exten => 501,n(LoopEnd),EndWhile

I don't understand, i do have the same thing you wrote above.

 
> Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
> Verbosity is at least 3
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Executing [...@phones:1] Answer("SIP/301-00000248", "") in new stack
>     -- Executing [...@phones:2] Wait("SIP/301-00000248", "2") in new stack
>     -- Executing [...@phones:3] Playback("SIP/301-00000248", 
> "velkommen_abacus") in new stack
>     -- <SIP/301-00000248> Playing 'velkommen_abacus.slin' (language 'en')
>     -- Executing [...@phones:4] Set("SIP/301-00000248", "Loop=0") in new stack
>     -- Executing [...@phones:5] While("SIP/301-00000248", "1") in new stack
>     -- Executing [...@phones:6] BackGround("SIP/301-00000248", 
> "tast123vent_") in new stack
>     -- <SIP/301-00000248> Playing 'tast123vent_.slin' (language 'en')
>     -- Executing [...@phones:7] WaitExten("SIP/301-00000248", "5") in new 
> stack
>     -- Timeout on SIP/301-00000248, continuing...
>     -- Executing [...@phones:8] Set("SIP/301-00000248", "Loop=1") in new stack
> [Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No 
> application '' for extension (phones, 501, 9)

You put '(LoopEnd)' in the place for the application. Hence empty
application with 'LoopEnd' as its input.

>   == Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-00000248'

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.co...@xorcom.com
+972-50-7952406           mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to