[asterisk-users] FreePBX, Asterisk and Twinkle - Testing a new setup

James B. Byrne byrnejb at harte-lyne.ca
Mon Apr 1 11:01:50 CDT 2013


I am experimenting with Asterisk having downloaded and installed the
FreePBX i386 CentOS-6.3 based distro and updated it. The current
package level on this system is:

asterisk11-11.3.0-49_centos6
freepbx-2.11.0beta2-112

I am using twinkle-1.4.2-7.el6 as a softphone testing tool.

There is no firewall on the asterisk host and SELinux is disabled on
it.  Fail2Ban is installed but I have made no alterations to the
default configuration, whatever it is.

The asterisk host is configured as 192.168.6.122.  The softphone is
configured on a separate host with a routable IP on our 216.xxx.xxx/24
netblock.  Both networks pass though an internal switch and are
firewalled from the outside world by a centos-6.4 based gateway host
using IPTables.  I have no difficulty in connecting to the asterisk
host either by ssh or by https.

I have initialised the FreePBX config and have selected the
user/device approach as this seems to fit our firm's employee
requirements more closely than the extension based configuration.  We
have several employees who frequently telecommute.

For the purposes of testing I have created two users, 11 and 12.  I
have configured a twinkle user profile for user 12.  I can place a
call to user 11 from twinkle and I get the IVR message for 'the number
you have called is not in service'.

I have tried to register Twinkle and this always fails. If I do :
# asterisk -rvvvvvvvv
CLI> sip show peers
Name/username             Host                                    Dyn
Forcerport ACL Port     Status      Description
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
offline]
CLI> sip show users
Username                   Secret           Accountcode     
Def.Context      ACL  ForcerPort

Which seems to say to me that I have nothing configured albeit I have
tried to through FreePBX.

At this point I am not trying to get a call out to our PTSN, although
I have the FXO port plugged into a live analogue line. What I am
trying to understand is the relationship between asterisk devices and
users.

The twinkle softphone has two lines (1 and 2).  It seems to me that I
should be able to configure each line as a separate extension and to
call one from the other.  What I cannot seem to discover is how to do
it.

Is it possible to do this?  How is it done?

-- 
***          E-Mail is NOT a SECURE channel          ***
James B. Byrne                mailto:ByrneJB at Harte-Lyne.ca
Harte & Lyne Limited          http://www.harte-lyne.ca
9 Brockley Drive              vox: +1 905 561 1241
Hamilton, Ontario             fax: +1 905 561 0757
Canada  L8E 3C3




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