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
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