Good catch. Unfortunately, I actually did have it in there as dialGSM, I just copied from the wrong version of the file when I copied and pasted it here.
This is what I get from sip show peers: Name/Username: IMSI262422146099205 Host: (Unspecified) Dyn: D Forceport: 0 ACL: Port: Unmonitored Status ... same for the other IMSI... 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Jacob On Tue, May 29, 2012 at 5:25 PM, James Thomas <jthomas...@gmail.com> wrote: > 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 -- _____________________________________________________________________ -- 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