[asterisk-users] Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not

[Vieri]

Hi,

I followed the how-to from
http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840

All works fine except for Asterisk->Alcatel calls.
Actually, calls from Asterisk to analog extensions on
the Alcatel work.
However, calls from Aserisk to digital extensions on
the Alcatel 4400 do NOT work.
I get this error in the Asterisk log:

    -- Executing Dial("SIP/4053-0823dd48",
"Zap/g1/2001|30") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/2001
    -- Channel 0/1, span 1 got hangup, cause 1
    -- Channel 0/1, span 1 received AOC-E charging 0
units
    -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/4053-0823dd48", "") in
new stack

It's odd that analog phones work whereas digital
phones don't.
I have telnet access to the Alcatel 4400 but am pretty
novice with its configuration settings.
However, all the steps in the how-to have been
completed.
Could it be related to the "Requested transfer
capability: 0x00 - SPEECH". Can that be adjusted?

A pri debug shows this:

    -- Called g1/3034
1 < Protocol Discriminator: Q.931 (8)  len=10
1 < Call Ref: len= 2 (reference 3/0x3) (Terminator)
1 < Message type: RELEASE COMPLETE (90)
1 < [08 03 81 81 80]
1 < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: Private network serving
the local user (1)
1 <                  Ext: 1  Cause: Unallocated
(unassigned) number (1), class = Normal Event (0) ]
1 <              Cause data 1: 80 (128)
1 -- Processing IE 8 (cs0, Cause)

What can "unallocated (unassigned) number" mean
exactly so that I can look for that in the Alcatel
system.

Thanks for any clues you may suggest.



       
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