Scott Huang
2012-Dec-20 16:15 UTC
[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' -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121221/0f45de00/attachment.htm> -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk sip_debug_on Type: application/octet-stream Size: 4673 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121221/0f45de00/attachment.obj>
Eric Wieling
2012-Dec-20 20:19 UTC
[asterisk-users] sip call failed in openbts with asterisk
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'