[Asterisk-Users] Call go on hold for no reason

[Chris Mason (Lists)]

I am trying to use ChanIsAvail to detect the best route for a call. I am 
testing by dialing an extension that is then forwarded to the DID.

Normally it will be an incoming PSTN call that is forwarded.

When I try it, I get put on hold for a few seconds and miss the 
beginning of the recorded message. Any ideas what is going on?

    -- Executing ChanIsAvail("SIP/501-304d", "IAX2/teliax&IAX2/voxee") 
in new stack
    -- Hungup 'IAX2/teliax-3'
    -- Executing Set("SIP/501-304d", "CH=IAX2/teliax") in new stack
    -- Executing NoOp("SIP/501-304d", "AVAILCHAN= IAX2/teliax") in new stack
    -- Executing Dial("SIP/501-304d", "IAX2/teliax/12222222222|60|tr") 
in new stack
    -- Called teliax/19032303776
    -- Call accepted by 207.174.202.4 (format g729)
    -- Format for call is g729
    -- IAX2/teliax-7 is making progress passing it to SIP/501-304d
    -- IAX2/teliax-7 answered SIP/501-304d
    -- Started music on hold, class 'default', on channel 'SIP/501-304d'
    -- Playing 'pbx-transfer' (language 'en')
    -- Stopped music on hold on SIP/501-304d
    -- Unable to find extension '' in context 'default'
    -- Playing 'pbx-invalid' (language 'en')
    -- Hungup 'IAX2/teliax-7'

My config is


exten => 700,1,ChanIsAvail(IAX2/teliax&IAX2/voxee)
exten => 700,n,Set(CH=${CUT(AVAILCHAN|-|1)})      ; New function style
exten => 700,n,NoOp(AVAILCHAN= ${CH})
exten => 700,n,Dial(${CH}/${DID},60,tr)
exten => 700,n,Goto(s-${DIALSTATUS},1)                            ; Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => 700,n,Hangup
;
exten => 700-CHANUNAVAIL,1,NoOp(Jump to NuFone)
exten => 700-CHANUNAVAIL,n,Dial(IAX2/${NUFONE}/${DID},60,tr)
exten => 700-CHANUNAVAIL,n,Hangup








-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: netconcepts_anguilla at yahoo.com 


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