[GUIDE] PBX in a Flash (PIAF) on Amazon EC2 with Free GV calling + SILK codec

PIAF (PBX in a flash) + AWS (Amazon Web Service) EC2

If you are making free GV calls using data, then you probably know that setting up an asterisk server is the best but also quite complicated. Thankfully, it just got a little easier because the PIAF team made the PIAF image available on AWS EC2 (a cloud based virtual machine).

Advantages over other asterisk/pbxes methods:
1. very small audio lag. Amazon servers have less latency than any other servers.
2. Since Amazon hosts your server, you don't pay any electricity cost. FYI, a PC running 24/7 costs about $15-20 a month.
3. The PIAF image (AMI) is already optimized: 1000MHZ + pre-installed asterisk and others. If you have used PIAF before, the whole setup process will take less than 30 minutes to complete.
4. You can create a backup image at any time and restore it later.

Disadvantages:
1. EC2 is only free for the 1st year. See the terms carefully to understand what is truly free. Use http://mikekhristo.com/ec2-ondemand-vs-reserved-instance-savings-calculator/ to estimate the cost after the 1st year. After the 1st free year is up, you can port the image to another AWS account.
2. Added security risk. Since EC2 is not in your network, it needs to open 22 (SSH), 80 (WEB) ports for you to access the server to configure. To overcome this issue, restrict who can gain access by known IP addresses from AWS.
3. In addition to some linux knowledge (PIAF runs on CentOS), you also need to learn how AWS works. Personally, I consider this as a plus.


Instructions (estimated minutes to complete in parenthesis, if you have used PIAF before):

1. Launch PIAF image in AWS EC2. (10 minutes)
Detailed instruction: http://nerdvittles.com/?p=5060
updated: there are 2 versions of PIAF: purple uses asterisk 1.8 and green uses asterisk 11 (beta). I am using PIAF-Green which supports silk codec.

2. Configure PIAF using Web GUI. (10 minutes)
Once PIAF is up and running, from your web browser, enter the Elastic IP address (given by AWS) to access web GUI.

2.1. Settings>Asterisk SIP settings>NAT settings. Click "Auto Configure" It will fill out the IP addresses. If you reboot, localnet address may change. You may use "10.0.0.0/8" instead.
* Whenever you make a change in Web GUI, scroll down and click submit. Then "Apply Config" red button will appear at the top. Click it occasionally to reload newly submitted configurations.
2.2. Settings>Asterisk SIP settings>Audio codecs. Select the codecs and reorder. (SILK codec will not show up here.)
2.3. Applications>Extensions. Add new SIP extensions. User extension and the secret are the username and the password you will use in your sip client to register with PIAF. Select nat=yes.
2.4. Other>Google Voice. Fill out your GV information. Asterisk must be restarted to take it into effect. In Web GUI, Admin>Asterisk CLI, execute "core restart gracefully" Once restarted, you can start making outbound GV calls from a registered phone. (Alternatively, you can use "simonics GVGW" service and add it as a sip trunk.)
2.5. Connectivity>Inbound routes. Add your GV number as DID number. Scroll down and set destination as your extension you created in #2.3.
2.6. From Google Voice webpage, confirm that Google Voice forwards all the incoming calls to gchat.
2.7. Register your sip client (such as csipsimple) with your PIAF server. The server address is the "elastic IP" address assigned by AWS.

Test in/out calls before going to the next step.

3. (optional) Add G729 codec. (5 minutes) (See post #2 for adding silk codec, which I'm now using for both wifi/3g.)
G729 codec is one of the best codecs for mobile connection. SSH to your PIAF EC2. In linux CLI, execute


To check whether you added G729 properly, execute in asterisk CLI


Look for G729 line and check they are not blanks.
Source: http://asterisk.hosting.lv/

4. (optional) Enable TCP transport to save phone battery. (5 minutes)
In my informal testing, TCP battery consumption is half of UDP's.

From Web GUI, asterisk SIP settings>Other SIP settings, add the following 2 lines.


Go Applications>Extensions and change transport to "TCP."

Next, you need to open TCP port 5060 in both EC2 and linux. For EC2, modify the security group. For linux, SSH to your PIAF


add the following line and save (ctrl-o) and exit (ctrl-x).

restart iptables in linux CLI

Use TCP transport in your sip client to connect to PIAF.



Tips/Tricks and Troubleshooting will be in the next posts.

If you couldn't understand the installation steps, check out a fellow's post. He provided several installation tips. http://xdaforums.com/showpost.php?p=41412908&postcount=184

Tips/Tricks

Tips/Tricks

1. Use ring group to ring all the extensions, if you have multiple sip phones.

2. Add more GV numbers
My family have 3 GV numbers + 6 phones. I added all 3 gtalk trunks to PIAF. OTOH, pbxes doesn't allow you to add more than 1 gtalk trunk. Since all 6 phones are in the same PBX, I can call my wife's phone by her extension number (or her ring group) instead of her GV number. If you do this, you need to modify outbound routes such that certain extensions use specific GV trunk. Otherwise, all the phones will use the first GV trunk to make outgoing calls and display the same CID.

3. Call forward to external phones.
In the ring group, you can have PIAF call regular phone number such as your cell phone. Just list the phone number followed by #. One problem is gtalk trunk doesn't allow CID spoofing so the external phone will show your GV# as CID.

4. Delayed call forwarding
Create 2 ring groups and put the number for delay in the 2nd group. Set 1st ring group's destination if no answer as the 2nd group and specify the ring time. I do this for my office phone because it goes to voicemail in 20 seconds (before GV voicemail answers). Now my office phone starts ringing after 7 seconds so it will never answer before GV voicemail does. You can also use this strategy with your cell phone such that your cell number rings after you have a chance to answer internet calls.

5. Secure your PIAF
If you followed pbxinaflash.com guide, then you opened port 80 and 22 to the world. Hackers can brute force using username = root. You should limit who can access these ports by IP address. Additionally, you can disable root SSH login and create a sudoer.
http://www.howtogeek.com/howto/linux/security-tip-disable-root-ssh-login-on-linux/
http://wiki.centos.org/TipsAndTricks/BecomingRoot

6. Add SILK codec
If you are on PIAF-green (asterisk 11), then you can add SILK codec. This is the codec that Skype uses so you may get the same excellent quality as Skype. From linux CLI,


Restart asterisk and from asterisk CLI, core show codecs and confirm that 4 silk codecs (silk8, silk12, silk16, silk24) are added. If not, read http://xdaforums.com/showpost.php?p=38398737&postcount=71.
To use SILK codec, From extension, set

disallow=all
allow=silk12

7. Eliminate echo
ryuker has posted a method to eliminate the echo.
http://xdaforums.com/showpost.php?p=42740388&postcount=306

Troubleshooting

There are many many reasons why GV calling fails. This means you will probably not get any help, unless you provide more information such as asterisk log. If you want to troubleshoot by yourself, try these first.

1. csip / piaf registration: Try a different sip client or use UDP connection.
2. GV: See whether you can make GV calls using GrooveIP. GV calling may not work for new accounts until you initiate a call from gmail.
3. Restart asterisk. If you add a GV trunk, it must be restarted to take into effect.
2. piaf / GV link: Add simonics GVGW as a sip trunk. https://simonics.com/gvgw/

If you have been following my iLBC thread, then here's a comparison between the 2 methods (PIAF vs iLBC, hereafter).

1. PIAF supports more codecs such as G722 and G729.
2. Less latency issue because of less call routing. PIAF route: caller>GV>PIAF>phone. iLBC method involves caller>GV>callcentric>pbxes>phone.
3. You can add more than 1 GV trunk w/ PIAF. pbxes only allows 1. This is handy, if your family use several GV numbers.

whahn1983 said:

I had tried a setup and got through all the nerdvittles guides setting up piaf purple with travelinman 3. Everything worked great with csipsimple when I was on WiFi. As soon as I disconnected from wifi and got on the cell network I could not connect. I ensured my phone ip was in the iptables and all the proper ports were forwarded through the router. I used dyndns for both my router and phone to ensure that ip tables had the right ip addresses. Never figured out the problem. If anyone has any ideas for what the problem could be I'll give it another try.

Sent from my Nexus 4 using xda premium

Click to expand...

Click to collapse

I think using dyndns for your phone is pointless and can actually pose a great security risk. If you are on Tmobile 3g, your phone is behind NAT. This means the IP of your phone actually belongs to the Tmo's router and not unique to your phone.

So including that IP address in iptables is a huge security risk. All the Tmo phones sharing the same IP address with your phone will have full access to your PBX.

The biggest challenge that you are facing is the fact that both your PBX and your phone are behind 2 different NATs. I know it's crazy. My guess is NAT settings. In nerdvittle's guide, there's an instruction on how to give access to remote clients. In addition, here is another good guide: http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension

Here are the instructions on how to enable TCP and change TCP port from default 5060 to something other such as 5055.

1. Settings>Asterisk SIP settings>Other SIP settings
tcpenable = yes
tcpbindaddr = 0.0.0.0:5055
2. Applications>Extensions>your extension, transport = TCP only.
3. From your router, port forward 5055 to PBX machine.
4. From PBX linux command, adjust firewall settings
iptables -A INPUT -p tcp -m tcp –dport 5055 -j ACCEPT
(You can also do this from PIAF linux webmin.)
5. From sip clients, server/proxy = your IP:5055.

Another issue that I found.

Can't use G722 codec over TCP when registered remotely. G722 works if UDP or within LAN.

Not sure whether this is a csip specific issue because I don't know other free sip clients with G722 codec.

osi13 said:

You should update the thread. TCP can be used on Asterisk 1.8, and PIAF on 1.8 is very stable.

Click to expand...

Click to collapse

I didn't say TCP cannot be used in 1.8 in OP. In fact, I have no doubt asterisk 1.8 supports TCP.

I was stating your guide wasn't written for TCP. Let me know if you have updated your guide with TCP.

If there's no hangup issues and G722 works with 1.8 when TCP/remote, then I'll ditch 11 in a heartbeat.

bluespire said:

Yeah, I always kept wondering why no one mentioned it as a solution. I don't know if it will solve the battery problem, as that may still be problem with using UDP itself. I suppose changing the timeout on UDP to something really high would work, since the VPN will maintain the TCP connection. On the other hand, you could also run the VPN on UDP, and I would bet that it would be better at staying connected than trusting CSS to keep its connection.

What the VPN does do really well, is eliminate all issue of allowing the PBX to face the internet. Just force it to accept connection only on its VPN port, and require a key file. Done.

Click to expand...

Click to collapse

I haven't had time to test this. But if my understanding of sip technology is correct, VPN will not solve UDP battery drain issue. For example, even if I use UDP within LAN, I have to set KA interval = 40 or qualifyfreq = 60. Otherwise, my phone will lose connection to the sip server until the next registration (which is less frequent than KA intervals).

errorcod3 said:

I was using my rpi for awhile on my home net. I switched to an amazon ec2 micro instance. much better speeds, reliability, and quality.

Click to expand...

Click to collapse

1. Is AWS EC2 behind NAT?

2. Anyone has an updated guide for installing asterisk 1.8 or 11 on EC2.