Re: [asterisk-users] Automatic attendant - Error in CLI.
On Fri, Jun 18, 2010 at 10:51:40AM +0200, Aksel Celasun wrote:
Hello dear list.
I am currently working on a Automatic attendant, and the core things work,
but I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The setup here is when 301 calls 501, and 301 doesn't enter an extension, it
will go in loop, 3 times, and then hangup...Can't get that working.
Could someone please help me?
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
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
CLI Output
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/301-0248, ) in new stack
-- Executing [...@phones:2] Wait(SIP/301-0248, 2) in new stack
-- Executing [...@phones:3] Playback(SIP/301-0248,
velkommen_abacus) in new stack
-- SIP/301-0248 Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set(SIP/301-0248, Loop=0) in new stack
-- Executing [...@phones:5] While(SIP/301-0248, 1) in new stack
-- Executing [...@phones:6] BackGround(SIP/301-0248,
tast123vent_) in new stack
-- SIP/301-0248 Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten(SIP/301-0248, 5) in new
stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set(SIP/301-0248, 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/301-0248'
--
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
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Re: [asterisk-users] Automatic attendant - Error in CLI.
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/301-0248, ) in new stack
-- Executing [...@phones:2] Wait(SIP/301-0248, 2) in new stack
-- Executing [...@phones:3] Playback(SIP/301-0248,
velkommen_abacus) in new stack
-- SIP/301-0248 Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set(SIP/301-0248, Loop=0) in new stack
-- Executing [...@phones:5] While(SIP/301-0248, 1) in new stack
-- Executing [...@phones:6] BackGround(SIP/301-0248,
tast123vent_) in new stack
-- SIP/301-0248 Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten(SIP/301-0248, 5) in new
stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set(SIP/301-0248, 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/301-0248'
--
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
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Re: [asterisk-users] Automatic attendant - Error in CLI.
Aksel Celasun wrote: This should be: exten = 501,n(LoopEnd),EndWhile I don't understand, i do have the same thing you wrote above. The difference between yours and his is that you had a n,(LoopEnd) and it should be n(LoopEnd) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] Automatic attendant - Error in CLI.
Ah, I missed the comma, thank you, and thank you Tzafrir Cohen! Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Doug Lytle Sendt: 18. juni 2010 11:41 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Automatic attendant - Error in CLI. Aksel Celasun wrote: This should be: exten = 501,n(LoopEnd),EndWhile I don't understand, i do have the same thing you wrote above. The difference between yours and his is that you had a n,(LoopEnd) and it should be n(LoopEnd) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
