[asterisk-users] sip call failed in openbts with asterisk

[Eric Wieling]

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Huang
Sent: Thursday, December 20, 2012 11:16 AM
To: asterisk-users at 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 at phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '3862c8d23be16ce36e564c3251cbc10c at 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'