Asterisk Voip to Voip "Bridge" (Webcallback)

Fri 29 October 2010 | tags: asterisk, bridge, callback, sip, sip to sip, voip, voip to voip, webcallback, -- (permalink)

Call it what you like. Sip to Sip, VoIP to VoIP, VoIP Bridge, Callback, Webcallback. The idea is to have a server somewhere call two SIP(VoIP) devices and connect the 2 calls together.

What's the point of it? Maybe you are stuck somewhere with a phone that only accepts incoming calls, and you want to make cheap voip calls. Or you are overseas, and it's cheaper to call from Australia to your location that from your location to Australia. For example you may have a payphone that accepts incoming calls but outgoing calls are expensive. Or you just want to enjoy nice cheap VoIP rates but don't have any VoIP hardware at your location, or poor dodgy internet, but have a mobile phone or landline.

What ever the reason, being able to have a VoIP call that connects to your location, then calls another party is a great thing!

Now before I go much further, if all you want is this ability to be called, and then have your remote party called, PennyTel has this Webcallback feature built in. Mynetfone doesn't, but if you can make outgoing calls you can use their ANI callback feature (you call a special number, your caller ID needs to not be blocked, and then you hangup after it rings, then it calls you back and lets you make your VoIP call). So, if you aren't going to be using this much, and don't need lots of customisation or special features, just signup for PennyTel and start using their Webcallback feature. Remember, you pay the cost of 2 calls using this feature, not 1. First, the cost to your number, then the cost to whom you are calling. So if both parties are mobile phones, then it certainly might not be cheap anymore. How ever, if your number is a landline in australia, and so is the party you are calling, then it'll cost a total of \$0.16 to make your call (on their Freedom Untimed plan). This is pretty cheap compared to most normal telco's.

Read on however, if you want the ability to use just about any VoIP provider, and want to maybe use multiple VoIP providers and have heaps of control over it all.

Asterisk, the VoIP/PBX mega funky software, will turn a normal server into a PBX/VoIP Gateway etc etc. When I first looked to a solution to this issue a few years ago, I looked at the task of setting up Asterisk and found it daunting, so gave up. PennyTel Webcallback essentially delivered what I needed so I left it alone until now. I now have Asterisk setup as a simple VoIP system that allows me to make 2 calls (using multiple VoIP providers if I want) and join them together.

Why is it better than PennyTel Webcallback?
For me it's better because it can work out cheaper for me. PennyTel callback is \$0.16 total (\$0.08 per each call to landline). However, if I'm calling someone who is a MyNetFone customer, if  I was to call their VoIP number from a MyNetFone VoIP account, the call would be free. And if I'm on a MyNetFone monthly plan, then I could have a good number of free landline calls each month. My optimal solution is currently combining both MyNetFone and PennyTel. PennyTel to the normal landline, \$0.08, and MyNetFone to the MyNetFone VoIP numbers, \$0.00. So it's a total of \$0.08 for the whole call! So depending on your location, and your destination number, and your VoIP provider plans, you can get your calls cheaper than using PennyTel webcallback. And just the fact that you can use this Callback method with just about any VoIP provider, priceless!

So lets get onto the easy part, setting it all up. While at first this can look daunting, in reality, it's very simple. This is all based on an Ubuntu 10.04 box, using the CLI. Quick note before we start, what ever machine you set this up on, needs to have enough bandwidth for 2 calls at the same time. So if you are putting it on a home ADSL connection, check you have enough up and down bandwidth for this. My server is hosted out on the internet and has enough bandwidth. A big reason for me setting this up is lack of a decent internet connection from my home to be able to do normal VoIP.

sudo apt-get install asterisk

Yes,this does pull in a far few bits of software, some of which isn't needed and I've not even poked at yet, but this is what we do to get Asterisk installed. Easy.

Now the setup. You need to edit /etc/asterisk/sip.conf to add your VoIP providers. Some providers give you the Asterisk details on their site, many others are just a google away, generally on

So drop down to the bottom of the [general] section in /etc/asterisk/sip.conf and add in a register line for each VoIP provider. Mine is in the following format. You can see that the username is repeated a few times on the mynetfone one. It can be different for other providers.

register =>

register =>

Now drop to the bottom of the /etc/asterisk/sip.conf file, and add in your "trunks" for the VoIP providers. In my situation, I don't want incoming calls (as then they need to be routed), so I just have the outgoing trunks.



Again, these can look different. A bit of googling around should help, or if you work from the above you should be able to work out the settings for your own provider. Note, username is no more in newer Asterisk versions, it's now defaultuser, so you may need to change what you find on the net.

Have a look in /etc/asterisk/manager.conf, you'll see you need to create a file in /etc/asterisk/manager.d/ to add a user. You may wish to do this so you can use the astman tool later, which helps with disconnecting calls and monitoring things. (You can do this through the asterisk cli, how ever it may be a bit harder).

The format of the file you create is as follows. /etc/asterisk/manager.d/somename.conf

secret = password
read = system,call,log,verbose,command,agent,user,config
write = system,call.log,verbose,command,agent,user,config

This username and password isn't related to your VPS at all, it's what you'll use when logging in with Astman or other tools.

Now we get to the bit we've been waiting for, setting it up to connect 2 calls.
Edit /etc/asterisk/extensions.conf. Comment out the include => demo line from the [default] section. There is a section in the [default] that should sort out Hangup/Busy detection and the likes, so the extensions we are adding can be very simple, however, you can customise these very easily to do more than just connect the 2 calls.

exten => _0X.,1,Dial(SIP/mynetfone-out/${EXTEN})

exten => _0X.,1,Dial(SIP/pennytel-out/${EXTEN})

And we are done. Everything should be setup now. The default install of Asterisk has autoloading on in /etc/asterisk/modules.conf so the rest should just work.

After all those edits, restart asterisk.

sudo /etc/init.d/asterisk restart

Now we create our call file.

Channel: SIP/pennytel-out/YOURNUMBER
Context: DialOutMyNetFone
Priority: 1
Archive: Yes

Very simple file. The Channel line is what calls you, so replace pennytel-out with the trunk you wish to use to call you. (It can be the same as the trunk used later on, or different. In this example, it's different as I want the cheap call to my landline, and the free call to a mynetfone number). The Context is the extension name that we defined in /etc/asterisk/extensions.conf, and the Extension is the number that you wish to call. Both for YOURNUMBER and WHOTOCALLNUMBER you need to know the format your VoIP provider uses. Some will allow just a localised version, for example in australia I can just dial 075xxxxxx or 040 123 4567, and MyNetFone understands them. Some other VoIP providers might need 6175xxxxxx instead. Dialing internation, some just need the country code, others need a prefix like 0011, etc etc.

Now to actually place the call. There are a number of methods of doing this, the end result needs to be a call file placed in the /var/spool/asterisk/outgoing directory, owned by the asterisk user, and placed in a way that asterisk can't start reading it before the file is all there. For this reason, the mv command is recommended and I'll demonstrate. (I first copy the file so I can keep a number of pre setup files stored).

sudo chown asterisk
sudo mv /var/spool/asterisk/outgoing/

As soon as the file hits that directory, asterisk reads it and executes it. If everything goes right, within 10 seconds your phone should start ringing. Pick it up, it'll then start ringing the other party, finally, when they pickup you can start talking! Done.

Sometimes the hangup detection doesn't work properly. I've not yet worked out why or how to fix it, I believe there are a number of things to tweak regarding this. What I currently do is run 'astman localhost' (from the machine running asterisk), login with the details we setup earlier, and I can think select the call and press hangup if it doesn't hangup correctly.

For more details about Call files, check out

Of course, it's not always practical to ssh into your server just to make a phone call. I'll leave the task of setting up other methods of creating and moving the call files to the reader. Please though, don't give your web server user full sudo access without a password! I'd recommend ether a database with a cron runner script (running as the asterisk user), or something similar.

Last things. Check your firewall. You'll need to open some RTP ports and some SIP ports. A google will reveal more. Try to limit what you open though, I've not given any instructions for securing asterisk in this post, that also is for your further reading.

Fork me on GitHub