We have had an article on installing Asterisk 13 on Ubuntu, but we did not go trough actually connecting with softphone and making the call. The article have been only about server side setup. Now we are going to build upon this article and connect Linphone to Asterisk 13.
Prerequisites
- Two Ubuntu computers on the same network (or computer and a VM, in my case it will be two laptops)
- One of them has asterisk installed
Installing Linphone and configuring Asterisk
Installing linphone is easy, on both of your computers type
sudo apt install linphone
After this we need to setup Asterisk to accept new users. On Asterisk machine, type
cd /etc/asterisk/
Here we moved to the directory where all of the Asterisk configuration files are located. We need sip.conf, but before editing it we will back it up:
sudo cp sip.conf sip.conf.orig
If you look into the file, it is huge. Scrolling takes time, so we will need to prune it to exclude all the commented lines, white spaces and only leave the lines that count as configuration.
So we open the file with vi while we are inside /etc/asterisk/
sudo vi sip.conf
And in vi we type this to remove comments
:g/^\s*;/d
Then this to remove white spaces.
:g/^$/d
What remains is the actual configuration. In the end of that configuration we need to add two users like this:
[miki] type=friend context=phones allow=ulaw,alaw secret=12345678 host=dynamic [bobby] type=friend context=phones allow=ulaw,alaw secret=87654321 host=dynamic
Next we enter the Asterisk with asterisk -rvvv command and in asterisk prompt we type
sip reload
And then to show peers
sip show peers
Connecting Linphone
On both computers, start linphone and enter the wizard to add users, with passwords that you put ins sip.conf
After this, we are not yet ready to call. We need to write a dial plan in extensions.conf. Lets first backup it and then make new empty file:
sudo mv extensions.conf extensions.conf.orig sudo nano extensions.conf
In empty file we will paste the dial plan.
[phones] exten => 100,1,NoOp(First Line) same => n,NoOp(Second Line) same => n,Dial(SIP/miki) same => n,Hangup exten => 200,1,NoOp(First Line) same => n,NoOp(Second Line) same => n,Dial(SIP/bobby) same => n,Hangup
Lets explain what all this means
In [] we put context, which is in our case phones. We operate on extensions, 100 for miki and 200 for boby. For doing things inside dial plan, we use things called applications. That are those things, like NoOp, Dial, and Hangup. Lets first explain the first line.
Exten is for extension, followed by => and then we say which number of the extension we operate on. It is 100, followed by 1 for the number of the line. NoOp is application that we use for the first line. In the second line things look differently but it is basically the same. Instead of exten we write same, because we are operating on same extension, 100.
This means that number 100 does not need to be repeated in second line and we move to line number but here we have a twist also. Instead of writing 2 (which we could also do), we will write n meaning next. There is simple reason for this. If we wrote 2 and then inserted another line between 1 and 2 with new application, we would need to change all line numbers. This way, if we write n, we do not need to do this.
When we move to second extension we are at the square one again. First line needs to be written in full, and while cards can be used for rest.
Making first call
After we have made a dialplan and saved it we are ready to call. Before we do that, lets first reload dialplan. That is mandatory step whenever we add new configuration.
asterisk -rvvv
After entering prompt
dialplan reload
Lets enter asterisk and look at our peers now
ThinkPad-X220T*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description bobby/bobby 192.168.1.6 D Auto (No) No 5060 Unmonitored miki/miki 192.168.1.5 D Auto (No) No 5060 Unmonitored
We have two peers connected, and we can make a call from one to another. Lets call bobby
== Using SIP RTP CoS mark 5
-- Executing [200@phones:1] NoOp("SIP/miki-0000000a", "First Line") in new stack
-- Executing [200@phones:2] NoOp("SIP/miki-0000000a", "Second Line") in new stack
-- Executing [200@phones:3] Dial("SIP/miki-0000000a", "SIP/bobby") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/bobby
-- SIP/bobby-0000000b is ringing
As we can see, the calling works.
How to find more applications
To write more complex dial plans, you would need to use more applications. You can browse applications in Asterisk prompt in several ways. For example you want dial applications, you would type core show applications dial
ThinkPad-X220T*CLI> core show applications like dial -= Matching Asterisk Applications =- Dial: Attempt to connect to another device or endpoint and bridge the call. RetryDial: Place a call, retrying on failure allowing an optional exit extension. -= 2 Applications Matching =-
It gives us two applications, one we already used and one is new to us, RetryDial. So if you type
core show application RetryDial
(notice no s, just application followed by the name), you will get the lengthy help file for application you selected.
But suppose you don't like either of those two apps, you want something more. You can expand your search by adding more vague definition like this
core show applications describing dial
I got 41 app by this search.
Conclusion
So building on our previous Asterisk article, we have shown how to configure Asterisk for use with Linphone. We have made a dialplan and made first call over our Asterisk server. Also we have introduced how to find applications you need to write your desired dial plan. Thank you for reading, and have fun with telephone calling.
