Re: [asterisk-users] unable to create channel of type 'SIP'
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 da...@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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 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/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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asterisk-users
[asterisk-users] unable to create channel of type 'SIP'
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
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 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)
--
_
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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
