Cause 20 means your SIP device is not registered or you do not have an IP specified for it in your peer.
"sip show peers" will show that. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Huang Sent: Thursday, December 20, 2012 11:16 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip call failed in openbts with asterisk Hi I met a problem in asterisk, please see message in the following, the detail debug log is in the attached file. can someone help to point out where to correctly configure asterisk, thanks a lot ! BR/Scott -------> -- Executing [8690@phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '3862c8d23be16ce36e564c3251cbc10c@127.0.1.1:5060' Method: INVITE [Dec 21 00:05:39] WARNING[2838]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466990004244439-00000014' status is 'CHANUNAVAIL' -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
