One such SIP compliant softphone is called 3CXPhone. This is a simple client that will work over both WiFI and 3G.

Bandwidth!! – One thing to consider when using a softphone on your Android handset is bandwidth. This might be a problem if you have a cap on your WiFi, but is much more likely to be an issue if you use 3G to make VOIP calls. Here are some quick facts and number to help you decide which codec to use -

G711 – good audio quality – uses around 72MB per hour    * also referred to as ulaw and alaw.

GSM – ok call quality – uses around 25MB per hour

G729 – ok call quality – uses around 21MB per hour    * not included with Asterisk or 3CXPhone

So above are 3 common codec choices with Asterisk. G711 has good quality audio but the trade-off is the amount of bandwidth it uses. If you have a small 3G cap you could quite quickly burn through it. The audio quality with GSM and G729 is OK. It’s perfectly understandable, and whether it’s acceptable or not will depend on the person making/receiving the call and their expectations.

G729 has the lowest bandwidth requirements of the 3 but it is not included with Asterisk by default (and is not available for 3CXPhone at all) as there are license costs to run it. You can add G729 to Asterisk but there is a license cost from Digium (there is an opensource version of G729 for Asterisk but you should be aware of any license restrictions - http://asterisk.hosting.lv/). You will also find that there is a higher cost for Android softphones that can run G729.

I would also recommend something like 3G Watchdog if you need to keep track of how much bandwidth you are using - https://play.google.com/store/apps/details?id=net.rgruet.android.g3watchdog&hl=en

Setup – Installation was straight forward. Just create an extension in FreePBX and then you just need the server IP (or name), extension number and extension secret in 3CXPhone. One thing I did notice was that disabling codecs in 3CXPhone did not seem to work for me. I disabled G711 in 3CXPhone and had G711 and GSM enabled in FreePBX for the extension. The call was still make using G711 and I don’t think that should be the case. If you want to use GSM I would just have GSM enabled for that extension in FreePBX. It would also be worth double checking what codec is used by checking a call in progress!

Making calls - 3CXPhone integrates with your Android contact list so it’s pretty easy to make calls. Just select the contact from the list, or dial the number. One thing to bear in mind is that the format the Asterisk system expects the phone number in may not be the same as your mobile operator. The interface is pretty simple with the ability to put a call on hold or transfer a call. One omission is that there is no call timer shown when a call is in progress. This would have been very useful if you’re trying to monitor bandwidth usage.  (in fact a SIP client with a built in bandwidth monitor would be a great idea …)

Receiving calls - Receiving calls worked OK. Just dial the extension number from another extension, or set the extension as a destination for an Inbound Route in FreePBX. One thing to bear in mind is that with power saving potentially turning off the WiFi or patchy 3G coverage when moving around is that receiving calls using any Android client will likely be a hit and miss affair.

Trouble with SIP - The ability to make inexpensive calls from a mobile phone over 3G has not been lost on the mobile phone carriers! Some actively block SIP connections so you may find that you’re not able to use it. Also, your terms of service may restrict your ability to make VOIP calls. It’s worth checking this before you find that your mobile phone carrier has cut you off!

Final thoughts - Using an Android SIP softphone is a great solution for some uses. If you’re by permanent WiFi that has a good signal and little interference then call quality can be great. If you’re using 3G then quality is certainly going to be hit and miss, probably unusable in a poor 3G signal area, or when being handed off between cell towers. Being able to use the internet to make inexpensive international calls is great. I’ve made calls from New Zealand to the UK using a SIP softphone over 3G and the call quality has been fantastic. It’s not always going to work that way, but when it does it’s amazing!

Another possibility  - If you make regular international calls to the same number then another option is to get a local DDI number and then forward that in FreePBX to the international number. For example I have a UK phone number that forwards to a land line in New Zealand. This way I can call a local UK number (included in my mobile plan minutes) that forwards to regular phone in New Zealand. International phone calls from your cell phone to the other side of the world for 1p per minute!  (plus the cost of the DDI number and any mobile phone plan minutes)

There are many stolen credit cards and PayPal accounts in circulation. You should absolutely not allow automatic signup/activation.

So, with that said, how do you turn on the customer signup page? In A2Billing go to SYSTEM SETTINGS / Global List / signup and set ‘enable_signup’ to ‘yes’

You should now be able to go to – https://vps_ip_address/signup.php

The easiest way to either delete or mass archive the recordings is via SCP. This works over the SSH port (22) and you use the root/ssh login details that were provided in the welcome e-mail.

If you’re a Windows user then there a nice, simple, free application called WinSCP - http://winscp.net/eng/index.php

Just enter your VPS IP address and the root username and password -

and then browse to /var/spool/asterisk/monitor where the recordings are stored -

from there you can either delete the “.wav” files directly or move them off your VPS and on to your PC

fail2ban
monitors Asterisk logs for failed ‘Register’ attempts and blocks the IP using IPTables. This means if you get yourself blocked it will appear as though the server has gone down

denyhosts
monitors /var/log/secure for failed SSH attempts and just blocks the IP for SSH access. You will get connection refused just for SSH if you get yourself blocked

It’s possible to whitelist your own IPs so that they don’t get accidentally blocked by following the instructions below.

You should replace 123.123.123.123 with your own IP -

It’s just a neat little script that grabs all of your Google contacts and automatically inserts them in to the Asterisk CallerID database.

Once the database is updated you can tell FreePBX to lookup the caller ID in the Asterisk database and present that with any incoming calls -

They are not the easiest boxes in the world to set up, with the web GUI being rather confusing. Eventually though the GoIP was configured correctly but still calls we intermittent, with connection to the Asterisk server being lost.

As the GoIP box was being hosted behind a residential ADSL router the issues were typical of NAT/Firewall problems. They were probably being caused by a mixture NAT and a SIP ALG (application layer gateway) in the ADSL router.

The solution was to use the new SysAdminMan VPN:PBX template which uses OpenVPN to create a secure VPN tunnel to the Asterisk server. The customer purchased a TP-Link 1043 router, flashed the SysAdminMan firmware, and the GoIP was connected to the VPS over the VPN. This meant that private IP address ranges were used, with no NAT happening at all. This is a great example of how using VPN:PBX to provide a VPN connection to your Asterisk server is easy to deploy and works around any NAT/Firewall issues.

Here is a diagram showing the customers setup -

See here for more information - http://sysadminman.net/sysadminman-vpnpbx-hosting.html

There is also a custom firmware (based on OpenWRT) available for the TP-Link TL-WR1043ND broadband router. Details of how to install this custom firmware can be found at - http://sysadminman.net/blog/2012/vpnpbx-getting-started-3890

While SysAdminMan does not sell the TP-Link TL-WR1043ND broadband router directly, it is available from many retailers including Amazon.

Here is an example of how VPN:PBX and the TL-WR1043ND could be deployed -

It can be deployed at a site and provide the following benefits -

• No changes required to existing router/firewall  (depending on deployment layout chosen)
• None of the problems of Network Address Translation associated with SIP as no NAT happens between the VOIP phones and the Asterisk server
• Voice traffic to the Asterisk server is encrypted
• Easy to deploy with custom firmware for the TL-WR1043ND router which includes OpenVPN client and GUI page for encryption settings
• Multiple TL-WR1043NDs can be deployed at multiple sites and connected to the same VPN:PBX server

Please see the product page for more information, or use the contact form if you have any questions.

While using FreePBX Endpoint Manager is possible with a normal SysAdminMan system, using it with VPN:PBX is really simple as no firewall or router changes are required.

Endpoint Manager is a FreePBX module that allows you to configure VOIP handsets centrally – no settings are required to be entered on the handset itself – just plug it in!

Here is the way it works -

1. The phone is switched on and given a DHCP address by the TP-Link TL-WR1043ND router
2. It is also given the IP address of the Asterisk server as it’s TFTP server – in this case of a SysAdminMan VPN:VPS that’s 10.98.0.1
3. The phone contacts the TFTP server (10.98.0.1) and asks for it’s configuration settings. It gives the server its MAC address to identify itself
4. EPM manager identifies the handset based on its MAC address and configures the phone

Brilliantly simple and also makes it very easy to deploy new phones and make changes to deployed phones. Endpoint Manager supports lots of different manufacturers handsets including – Aastra, Grandstream, Polycom, Snom, Cisco and Yealink.

So, how does it work in practice? Here is a quick guide for deploying a new handset with Endpoint Manager -

First get the MAC address of the handset. In this case an Aastra 6755i -

Next create an extension as normal in FreePBX. The only settings required are User Extension, Display Name and Secret -

Next enable the required handset template in Endpoint Manager / Endpoint Configuration -

Finally add a device in Endpoint Manager End Point Device List -

#

and clink on Rebuild Configs for All Phones -

Now when you connect your handset to the network it should connect to Endpoint Manager and download it’s configuration. Here’s mine making an echo test call -

Because the TP-Link router creates a OpenVPN tunnel to the VPN:PBX server no NAT, firewall or router changes are required.

It is designed to work with the TP-Link TL-WR1043ND broadband router. A custom OpenWRT firmware is provided to simplify the connection of your network to your SysAdminMan VPS, securely and without the hassle of NAT.

STEP 1 – DOWNLOAD THE CUSTOM FIRMWARE

Download the custom  TP-Link TL-WR1043ND firmware with OpenVPN client installed -

http://sysadminman.net/openwrt/tl-wr1043nd-sysadminman-v1.0-squashfs-factory.bin

STEP 2 – PLUG IN THE TP-LINK WR1043ND

1. Plug a PC/Laptop in to the yellow port 1
2. Plug the blue WAN port in to a spare port on your existing internet router

* While every effort is made to test the firmware provided no responsibility can be accepted if you use the firmware above and it renders your router unusable. This is always a risk that is taken when flashing any new firmware to a device *

You must log in from a computer connected to a yellow LAN port.

Default IP address : 192.168.1.1

Default username : admin

Default password : admin

Select System Tools / Firmware upgrade and then choose the firmware image you downloaded above -

Click upgrade and the new firmware will be loaded and the router restarted. Do not disconnect the power while this is happening!! -

STEP 4 – RESET YOUR COMPUTER OR RENEW YOUR DHCP LEASE

You will now need to get your computer to pick up a new DHCP address from the TP-Link TL-WR1043ND. This will be in the range 10.99.99.X. This can be done by either rebooting, pulling and reinserting the network cable or running the command to renew the IP address. Please note that the WiFi is disabled on the TP-Link TL-WR1043ND firmware so you must use a cabled connection.

SETEP 5 – LOG IN TO YOU TP-LINK ROUTER AGAIN

Default IP address : 10.99.99.1

Default username : root

Default password : sysadminman

STEP 6 – ENTER THE SYSADMINMAN VPS CONNECTION DETAILS IN TO YOUR TP-LINK ROUTER

For this step you will need the IP address of your SysAdminMan server plus the 3 files that will be sent to you separately. The 3 files are called  ca.crt, tplink.crt and tplink.key.

Click on Network / SysAdminMan and you will see the screen below. Enter your SysAdminMan VPS IP address in the box provided and upload the 3 files. Ensure you upload the correct file to the correct place. Then click Save & Apply -

Next reboot the router. There is an option to do this under System / Reboot -

Now if you log back in and select Network you should see that the ‘SYSADMINMAN’ interface has been assigned an IP address of 10.98.0.X -

STEP 7 – FINISHED!

And that should be everything. Now any device connected to your TP-Link should be given an IP address in the range 10.99.99.x and be able to connect to your SysAdminMan VPS server using the address 10.98.0.1 over the secure OpenVPN tunnel.

Also, any phone that is connected to your TP-Link that has auto provisioning set will be given a TFTP server address of 10.98.0.1, your SysAdminMan VPS address. This means that you can use the FreePBX EndPoint Manager module to configure your telephone handsets.

NETWORK DETAILS

• TP-Link LAN address : 10.99.99.X
• SysAdminMan VPS address : 10.98.0.1

Having servers based in the UK and US with great network connections has enabled us to provide stable VOIP platforms to customers in countries as far away as New Zealand, Spain, Canada and India!

Apparently though there are 196 countries – so still some room for growth yet!