[asterisk-users] PBX -> PRI -> * -> Telco not working

Mikel Lindsaar raasdnil at gmail.com
Sat Nov 15 12:09:36 CST 2008

OK, made some progress.
With some help from friends on #Asterisk, found that the NEC was doing post
connect dialing.

So then, added the following to extensions.conf:

exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Read(DialedNumber)
exten => s,n,Dial(DAHDI/g2/${DialedNumber},,T)

Which now captures the number and tries to dial, but then I get:

    -- Accepting call from '' to 's' on channel 0/28, span 1
    -- Executing [s at from-nec:1] Answer("DAHDI/28-1", "") in new stack
    -- Executing [s at from-nec:2] Set("DAHDI/28-1", "TIMEOUT(digit)=2") in new
    -- Digit timeout set to 2
    -- Executing [s at from-nec:3] Set("DAHDI/28-1", "TIMEOUT(response)=5") in
new stack
    -- Response timeout set to 5
    -- Executing [s at from-nec:4] Read("DAHDI/28-1", "DialedNumber") in new
    -- User entered '14140400400000'
    -- Executing [s at from-nec:5] Dial("DAHDI/28-1",
"DAHDI/g2/14140400400000,,T") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g2/14140400400000
    -- DAHDI/32-1 is proceeding passing it to DAHDI/28-1
    -- Channel 0/1, span 2 got hangup request, cause 31
[Nov 16 16:09:30] WARNING[5828]: app_dial.c:827 wait_for_answer: Unable to
forward voice or dtmf
    -- Hungup 'DAHDI/32-1'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'DAHDI/28-1' status is 'CHANUNAVAIL'
    -- Channel 0/28, span 1 got hangup request, cause 16
    -- Hungup 'DAHDI/28-1'


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081116/c44248bd/attachment.htm 

More information about the asterisk-users mailing list