(4/5) Asterisk – Making your first phone call (VoIP to VoIP ONLY)

(4/5) Asterisk – Making your first phone call (VoIP to VoIP ONLY)

July 18th, 2015 // 1:13 pm @


In the previous articles we covered everything from installing the Astreisk, to configuration, and then setting up your SIP accounts and Dialplans. Now, let’s get our phones set up and make the first test phone call. We will also talk about how to quickly debug our SIP connections


  1. Fully configured Asterisk Server
  2. Softphone phone
  3. VoIP Desk Phone (Optional)


We need at least two phones, we can use two softphones, but if you have any deskphones that supports VoIP, it will do it as well.

For softphone, I suggest Zoiper, it’s free and available on all platforms. For the sake of this articles, let’s install one on your laptop/PC and the other one on your Smartphone or another computer. However, there are plentty opf free applications that you can use.

You can download Zoiper here: http://www.zoiper.com/en/voip-softphone/download/zoiper3

NOTE: We are doing this example assuming that you are testing your Asterisk on a LAN, if you are trying to connect to your asterisk using WAN you need a public IP and also need to create Port forwarding on your firewall. See here for the complete list of Ports 

NOTE: From here on the settings apply to all VoIP phones, soft/desk phone.

Adding an account:

Let’s open Zoiper and add an account – (Settings > Create a new account > SIP)

you need the account info we created earlier, see here

For phone-1, we have:

Username: user-one
Password: A9ahui
Extension: 6020

so your user@host will be the username@ASTERISK_IP_ADDRESS

in our case: user-one@

then enter your password, and you can leave Domain empty for now!

For phone-2, we have:

Username: user-two
Password: a13UI98a
Extension: 3060

Repeat the steps above and configure the second phone.


If all goes well, you should see Zoiper logging in, and at the bottom, you should see the status “Online”

Let’s login to Asterisk CLI, and test our connections:

[root@localhost soap]# asterisk -r


You will see the Asterisk welcome message:

Asterisk 11.16.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
Connected to Asterisk 11.16.0 currently running on localhost (pid = 758)

now let’s see if our phones are connected successfully, go ahead, and issue the “sip show peers” command, you will see a list of accounts online.

localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
user-one D Yes Yes 55982 Unmonitored
user-two D Yes Yes 55954 Unmonitored
demo-dummy (Unspecified) D Yes Yes 0 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline]

As you can see above, I have the list of all SIP accounts in the Asterisk, however, there are only 2 online. I have created a 3rd account, so you can see the difference between online, and offline ones. The two that we registered have a valid IP address and also, have associated a port to them. However, the 3rd account is inside my sip.conf, but I didn’t connect it to a phone, so it’s just there, not doing anything. Now if I add that to the 3rd phone, and issue “sip show peers” again, I should see the IP and port for the 3rd phone.


If you don’t see your two accounts registered, something is wrong! Try few of these:

  • Make sure your user@host combination is correct
  • Make sure Asterisk is not behind a firewall
  • Make sure sip.conf compiles correctly, you can issue “sip reload” and you should see an OK message without an error
  • Do the same for dialplan and extensions.conf, issue “dialplan reload”
  • Double check the sip configuration, if it’s not correct, Asterisk won’t register your phone

Test Call

After a long wait, it’s time to test our phone, just dial the other phone’s extension, and you should get a ring! you can also try our test scenarios:

Try these:

  • One Way COMM test ( Ext: 1005 )
  • Two Way COMM test (Ext: 1001 )


Well, great job! now you have a fully functional Asterisk. I will add the Video over IP Capabilities, in a quick, side article.

Category : VoIP - telephony

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