[asterisk-users] unable to create channel of type 'SIP'

[Jacob Fenwick]

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 <jthomasdpu at 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.fenwick at 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)
>>
>> --
>> _____________________________________________________________________
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>
>
>
> --
> _____________________________________________________________________
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
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