[asterisk-users] Automatic attendant - Error in CLI.

[Aksel Celasun]

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
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 [501 at phones:1] Answer("SIP/301-00000248", "") in new stack
    -- Executing [501 at phones:2] Wait("SIP/301-00000248", "2") in new stack
    -- Executing [501 at phones:3] Playback("SIP/301-00000248", "velkommen_abacus") in new stack
    -- <SIP/301-00000248> Playing 'velkommen_abacus.slin' (language 'en')
    -- Executing [501 at phones:4] Set("SIP/301-00000248", "Loop=0") in new stack
    -- Executing [501 at phones:5] While("SIP/301-00000248", "1") in new stack
    -- Executing [501 at phones:6] BackGround("SIP/301-00000248", "tast123vent_") in new stack
    -- <SIP/301-00000248> Playing 'tast123vent_.slin' (language 'en')
    -- Executing [501 at phones:7] WaitExten("SIP/301-00000248", "5") in new stack
    -- Timeout on SIP/301-00000248, continuing...
    -- Executing [501 at 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)
  == Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-00000248'
asterisk*CLI>

sip.conf regarding sip 501

[501]
type=friend
secret=XXXXXX
host=dynamic
context=phones
mailbox=501 at default
callerid=Sentralbord
qualify=yes



Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
aksel at abacus-it.no<mailto:aksel at abacus-it.no>

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