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 2 channels in use out of the 16. When this happens, the user just needs to re-dial and the call goes through OK. [2008-10-14 15:41:40] -- Executing [EMAIL PROTECTED]: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 [EMAIL PROTECTED]: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 [EMAIL PROTECTED]: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 [EMAIL PROTECTED]: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' Could you please advise on what could be causing this? Thank you. Veselin _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users