Hi<br><br> 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><br>BR/Scott<br>
<br>-------><br> -- Executing [8690@phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack<br>Really destroying SIP dialog '<a href="http://3862c8d23be16ce36e564c3251cbc10c@127.0.1.1:5060">3862c8d23be16ce36e564c3251cbc10c@127.0.1.1:5060</a>' Method: INVITE<br>
[Dec 21 00:05:39] WARNING[2838]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<br> == Everyone is busy/congested at this time (1:0/0/1)<br> -- Auto fallthrough, channel 'SIP/IMSI466990004244439-00000014' status is 'CHANUNAVAIL'<br>
<br>