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Update README.rst

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lemueldsouza 2021-08-04 15:52:49 +05:30
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@ -48,6 +48,110 @@ You can monitor the FreePBX service with command::
systemctl status freepbx
Steps to setup a basic FreePBX configuration with a SIP extension
------------------------------------------------------------------
1. After installing PBX as part of IIAB, please visit http://box.lan:83/freepbx and proceed with initial configuration. You will be asked to setup your username and password the first time you login which will be used in future to login to the FreePBX configuration screen. Once you login, select the first option 'FreePBX Administrator'.
2. Change the default asterisk password
Go to Settings >> Asterisk settings. Click on 'Submit' button below and then clic'Apply config' that'll appear on the top right side of the web page.
3. Change asterisk SIP settings
Go to Settings >> Asterisk SIP settings >> Under NAT settings, clicking "Detect Network Settings" will populate your external IP
Under Local networks, enter your local IP settings in the form of IP/CIDR or IP/NETMASK such as, “192.168.0.0/24" or “192.168.0.0/255.255.255.0”
Click on 'Submit' button below and then click 'Apply config' that'll appear on the top right side of the web page.
Refer - https://wiki.freepbx.org/display/FPG/Asterisk+SIP+Settings+User+Guide
4. Create SIP phone extensions to enable you to make calls within your network
Go to Applications >> Extensions >> Add Extension >> New chan_pjsip extension
**Extension** - <<An extension number of your choice, like 101>>
**Display name** - <<Your name>>
**Secret** - <<Add a strong password here>>
Click on 'Submit' button below and then click 'Apply config' that'll appear on the top right side of the web page.
Using the same steps, you could create more extensions for other users.
5. Register the extension on your softphone app
You can now register these extensions using a softphone app on your smartphone. For this example we will use the Linphone app on an Android phone
Once you open the app, follow these steps
1. Select option "USE SIP ACCOUNT"
2. Enter the following details that you set in the FreePBX console
Username - 101
Password - Password you set for your extension
Domain - Asterisk server IP address (To find this out, on the system where you've installed FreePBX, go to Terminal and run 'ifconfig' to find your IP address)
3. Select "UDP" option under TRANSPORT
4. Click on login.
5. If connection is successful, you will see 'connected' with a green cirle on the next screen
6. Make a call to a random number or another extension you've created. You should be able to see activity on the applet at the right side of your FreePBX Dashboard
Refer - https://wiki.freepbx.org/display/FPG/Extensions+Module+-+PJSIP+Extension
Troubleshooting
----------------
1. Check if asterisk is up and running
Execute the command on your terminal and an asterisk console should open
sudo asterisk -rvvv
2. If you see a "Asterisk not connected" in red on the FreePBX web console, check if asterisk is 'running' using this command on your terminal
systemctl status asterisk
If asterisk is not running (status does not show 'running'), restart asterisk
sudo systemctl restart asterisk (confirm status shows up as running after executing this command)
3. If you see a "fwconsole read error" when you save settings, execute these commands on your terminal
sudo fwconsole chown
sudo fwconsole reload
4. Radcli error
In files /etc/asterisk/cdr.conf and /etc/asterisk/cel.conf, this line sometimes needs to be added: (possibly this manual step is no longer necessary with Asterisk 18.x now!)
radiuscfg => /etc/radcli/radiusclient.conf
In any case, make 100% sure the file /etc/radcli/radiusclient.conf is non-empty. You can end up with a zero-length file here, if IIAB's roles/pbx install was interrupted (it should be about 2-to-3 kBytes initially). Probably best to start over with a clean OS in such situations!
Also make sure any older lines including radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf are commented out within cdr.conf and cel.conf
Some useful asterisk commands and information
----------------------------------------------
1. pjsip show endpoints
This shows you the list of extensions along created on your FreePBX server along with its details
2. Asterisk log file is at /var/log/asterisk/full
3. If you do not see any activity on your asterisk console, you may need to increase the verbosity by executing either of these commands
core set verbose 3, OR
core set debug 3
4. To see all asterisk commands available
core show help
5. To see all commands that start with core show
``core show [tab]`` or ``core show?``
Raspberry Pi Known Issues
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