asterisk users - Oct 2008 - Asterisk 1.4: ISDN congestion warnings

Hello,
I'm using Asterisk with an ISDN30e PRI line (only 16 channels active).

Every now and then I get a CONGESTION error even-though there are only
1 or 2 channels in use out of the 16 at that time.

When this happens, the user just needs to re-dial and the call goes
through OK.

On a SNOM phone when the problem occurs, a "Service Unavailable 907"
error is shown.


[2008-10-14 15:41:40]   -- Executing [s at macro-to-isdn:1]
Dial("SIP/216-bc0aab90", "Zap/g1/0123456789") in new stack
[2008-10-14 15:41:40]   -- Requested transfer capability: 0x00 -SPEECH
[2008-10-14 15:41:40]   -- Called g1/0123456789
[2008-10-14 15:41:40]   -- Zap/1-1 is proceeding passing it to SIP/216-bc0aab90
[2008-10-14 15:41:41]   -- Channel 0/1, span 1 got hangup request, cause 34
[2008-10-14 15:41:41]   -- Zap/1-1 is circuit-busy
[2008-10-14 15:41:41]   -- Hungup 'Zap/1-1'
[2008-10-14 15:41:41]   == Everyone is busy/congested at this time (1:0/1/0)
[2008-10-14 15:41:41]   -- Executing [s at macro-to-isdn:2]
Goto("SIP/216-bc0aab90", "s-CONGESTION|1") in new stack
[2008-10-14 15:41:41]   -- Goto (macro-to-isdn,s-CONGESTION,1)
[2008-10-14 15:41:41]   -- Executing [s-CONGESTION at macro-to-isdn:1]
PlayTones("SIP/216-bc0aab90", "Busy") in new stack
[2008-10-14 15:41:41]   == Auto fallthrough, channel SIP/216-bc0aab90'
status is 'CONGESTION'
[2008-10-14 15:41:41]   -- Executing [h at internal:1]
Hangup("SIP/216-bc0aab90", "") in new stack
[2008-10-14 15:41:41]   == Spawn extension (internal, h, 1) exited non-zero on
'SIP/216-bc0aab90'


Any advise?

Thank you.
Veselin