Can you give me some pointers on where to read documentation on how to set up registered phones?
Also I'm wondering if maybe it would help if I tried setting up some softphones first. Can someone recommend some cheap softphones that work with asterisk? Jacob On Tue, May 29, 2012 at 5:36 PM, Danny Nicholas <[email protected]> wrote: > You can dial out from an unregistered SIP peer, but you can't receive a call > or call that peer. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Jacob Fenwick > Sent: Tuesday, May 29, 2012 4:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] unable to create channel of type 'SIP' > > 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 <[email protected]> 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 >> <[email protected]> >> 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/OpenBTSSettingUpAs >>> terisk 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 > > > -- > _____________________________________________________________________ > -- 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
