I think you need to change: exten => 2012,1,Macro(dialSIP,IMSI262428511722625) exten => 2013,1,Macro(dialSIP,IMSI262422146099205)
to: exten => 2012,1,Macro(dialGSM,IMSI262428511722625) exten => 2013,1,Macro(dialGSM,IMSI262422146099205) also what does sip show peers show, as opposed to sip show registry? On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick <jacob.fenw...@gmail.com>wrote: > 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 >
-- _____________________________________________________________________ -- 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