I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other.
I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create channel of type 'SIP' (cause 20 - unknown) If I type sip show registry it says there are 0 SIP registrations. Should I see both the phones registered at this point? If that's what's wrong, what am I doing wrong that's making the phones not able to register? Below is my Asterisk configuration. Jacob #/etc/asterisk/sip.conf [general] context=sip-external #... [IMSI262428511722625] callerid=2012 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info [IMSI262422146099205] callerid=2013 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info #/etc/asterisk/extensions.conf [macro-dialGSM] exten => s,1,Dial(SIP/${ARG1}) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-CANCEL,1,Hangup exten => s-NOANSWER,1,Hangup exten => s-BUSY,1,Busy(30) exten => s-CONGESTION,1,Congestion(30) exten => s-CHANUNAVAIL,1,playback(ss-noservice) exten => s-CANCEL,1,Hangup [sip-external] exten => 2012,1,Macro(dialSIP,IMSI262428511722625) exten => 2013,1,Macro(dialSIP,IMSI262422146099205) -- _____________________________________________________________________ -- 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
