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

[Jacob Fenwick]

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 <danny at debsinc.com> wrote:
> You can dial out from an unregistered SIP peer, but you can't receive a call
> or call that peer.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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 <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/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)
>>>
>>> --
>>> _____________________________________________________________________
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>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
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>>   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