Vieri
2007-May-25 04:19 UTC
[asterisk-users] Asterisk to Alcatel 4400 via PRI: analog extensions work - digital do not
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. ____________________________________________________________________________________Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469
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