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Zentrunk & Asterisk - Regular Trunking

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Overview

Zentrunk is a SIP Trunking service from Plivo that allows you to connect with fixed and mobile phones in over 200 countries. Connect your cloud or on-premise communication infrastructure to Plivo’s Zentrunk SIP Trunking service to connect to your customers easily.

This documentation provides a basic configuration to get Asterisk up and running with Plivo as the external SIP gateway.

To get started with Zentrunk using Asterisk you would need to do the following:

1. Install Asterisk on your environment.
2. Create a Trunk on Zentrunk using Plivo Console.
3. Configure an Outbound Trunk.
4. Configure the Inbound Trunk.

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Installation of Asterisk

For installing Asterisk, follow the instructions below:

• For Debian systems execute the following commands in the terminal.

  1. apt-get update
  2. apt-get upgrade
  3. apt-get install asterisk

• Follow the instructions given here to install Asterisk on CentOS.

• To install Asterisk on Windows, follow the instructions given here

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Create a Trunk on Zentrunk

You can create a trunk using Plivo Console. For more information on creating a Trunk on Plivo Console, see below links:

• Create an Outbound Trunk
• Create an Inbound Trunk

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Configuring Asterisk for Outbound Trunk

To configure the asterisk to connect to your Plivo Zentrunk, locate the root configuration of Asterisk on your machine. These locations vary from platform to platform.

In this case (Debian Jessie GNU/Linux System), the root configuration is present at /etc/asterisk/.

With the root configuration directory located, there are two major configurations that need to do -

1. Create a new SIP Channel
2. Create a Dial Plan.
3. Reload the configuration

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Step 1: SIP Channel

Create a new channel named “plivo-phone” at /etc/asterisk/sip.conf. This channel will be used in X-Lite to connect to asterisk. Also, create another channel called “plivo-trunk” which will connect to your Plivo Trunk.

[plivo-phone]
type=friend
context=Zentrunk
host=dynamic
secret=password1234

[zentrunk]
type=peer
context=Plivo
host=Termination SIP Domain of your Plivo Trunk
secret=Password for TestAuthGroup
username=Username for TestAuthGroup

A “plivo-phone” channel is created with the following attributes.

• Type=friend - Creates a user and peer connection
• Context=Zentrunk - Context is the identifier for a dialplan that will be loaded from extensions.conf. The sip.conf identifies and allows connections to the asterisk server. The context in the identifier allows the execution of call flow when a call is received from XLite.
• Host=dynamic - XLite can be connected from anywhere
• Secret=password1234 - Password to be used in X-Lite

The “zentrunk” channel is created with the following attributes.

• Type=peer - Creates a peer connection
• Context=Plivo - Context is the identifier for a dialplan that will be loaded from extensions.conf. The sip.conf identifies and allows connections to the asterisk server. The context in the identifier allows the execution of call flow when a call is received from XLite.
• Host=Termination SIP Domain of your Plivo Trunk
• Secret=Password for TestAuthGroup
• Username=Username for TestAuthGroup

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Step 2: Dialplan

Next, you should set up a Dial Plan. A Dial Plan tells Asterisk what to do when a call has to be placed. The “Content” attribute in the SIP channel connects a channel with a dialplan. Add a dialplan named “Zentrunk” in extensions.conf under /etc/asterisk/directory.

[Zentrunk]
exten => _1XXXX.,1,Set(CALLERID(all)="Your Plivo Number" <Your Plivo Number>)
exten => _1XXXX.,n,Dial(SIP/zentrunk/${EXTEN})
exten => _1XXXX.,n,Hangup()

The above dial plan has defined an extension for a number starting with the digit 1. When a call is made from X-Lite to a number that starts with 1, it hits the asterisk server first. The dialplan that satisfies this pattern matching get loaded, in this case, the above plan. The caller ID is set to your Plivo Number and the SIP INVITE is sent to “Zentrunk” channel which forwards the invite to Plivo Trunk from where the outbound call is placed.

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Step 3: Reload Configurations

1. Load the asterisk client.
2. Execute the following command in your terminal:
   Copy

   Ask AI

   asterisk -rvvvv
   

3. Reload the sip channel, and then execute the following command:
   Copy

   Ask AI

   sip reload
   

4. Reload the dialplan, and then execute the following command:
   Copy

   Ask AI

   dialplan reload.
   

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Configuring Asterisk for Inbound Trunk

To configure your asterisk to connect to your Plivo Zentrunk, locate the root configuration of Asterisk on your machine. These locations vary from platform to platform.

In this case (Debian Jessie GNU/Linux System), the root configuration is present at /etc/asterisk/.

With the root configuration directory located, there are three major configurations that you need to do -

1. Create a new SIP Channel
2. Create a Dial Plan.
3. Create a Sip Driver.
4. Reload the configuration

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Step 1: Create a new SIP Channel

Create a new channel named “6001” at /etc/asterisk/sip.conf. This channel will be used in X-Lite to connect to asterisk.

[general]
context=incoming

[6001]
type=friend
context=from-internal
host=dynamic
secret=1234
disallow=all
allow=ulaw

A 6001 channel is created with the following attributes.

• Type=friend - Creates a user and peer connection .
• Context=incoming - Context is the identifier for a dialplan that will be loaded from extensions.conf. The sip.conf identifies and allows connections to the asterisk server. The context in the identifier allows the execution of call flow when a call is received from XLite.
• Host=dynamic - XLite can be connected from anywhere
• Secret=1234 - Password to be used in X-Lite

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Step 2: Dialplan

Next, you should set up a Dial Plan. A Dial Plan tells Asterisk what to do when a call is received .The “Content” attribute in the SIP channel connects a channel with a dialplan. Add a dialplan named “incoming” in extensions.conf under /etc/asterisk/directory.

[incoming]
exten => _X.,1,Wait(1)
exten => _X.,n,Dial(SIP/6001)

The above dial plan has defined an extension for a SIP enpoint named 6001. When a call is made to your inbound number, it hits the Plivo first and then it is forwarded to your asterisk server .Once the dialplan is loaded and the call is placed to the soft phone registered as 6001 in your asterik

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Step 3: Create a SIP Driver

Create a new SIP driver named “6001” at /etc/asterisk/pjsip.conf with the below information.

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=6001
aors=6001

[6001]
type=auth
auth_type=userpass
password=1234
username=6001

[6001]
type=aor
max_contacts=1

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Step 4: Reload Configurations

1. Load the asterisk client.
2. Execute the following command in your terminal:
   Copy

   Ask AI

   asterisk -rvvvv
   

3. Reload the sip channel, and then execute the following command:
   Copy

   Ask AI

   sip reload
   

4. Reload the dialplan, and then execute the following command:
   Copy

   Ask AI

   dialplan reload
   

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Secure Trunking using chan_pjsip

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Overview

In this section, we will guide you through the steps to configure Asterisk to implement secure trunking for outbound calling. To configure the asterisk using chan_pjsip to connect to your Plivo Zentrunk, locate the root configuration of Asterisk on your machine. These locations vary from platform to platform. In this case (Debian Jessie GNU/Linux System), the root configuration is present at /etc/asterisk.

With the root configuration directory located, two major configurations need to be done-

1. Create an endpoint for Trunk
2. Create a Dial Plan
3. Reload the configuration / Restart Asterisk

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Step 1: Create an endpoint for Trunk

Create a new endpoint named zentrunk_endpoint_out at /etc/asterisk/pjsip.conf. You can use this endpoint to connect Zentrunk.

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pjsip.conf

Endpoint to connect Zentrunk

[zentrunk_endpoint_out]
type=endpoint
transport=transport-udp
aors=zentrunk_aor
disallow=all
allow=ulaw
outbound_auth=zentrunk_auth

[zentrunk_aor]
type=aor
contact=sip:xxxxxxxxxxxxxxxx.zt.plivo.com:5060

[zentrunk_auth]
type=auth
auth_type=userpass
password=mitesh123
username=mitesh

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

• [zentrunk_endpoint_out]: This is an endpoint definition.

• [zentrunk_aor]: This defines the Address of Record to be used by endpoint - zentrunk_endpoint_out. This tells Asterisk where an endpoint can be contacted. For that you will need to configure contact with the url which will point to Zentrunk as below:

  Copy

  Ask AI

  contact=sip:<"TERMINATION SIP DOMAIN" of your ZENTRUNK's Outbound Trunk> :5060
  

• [zentrunk_auth]: This defines authentication for zentrunk_endpoint_out. When the Trunk challenges for the INVITE from Asterisk, this section will be used to authenticate.

• [transport-udp]: The endpoint zentrunk_endpoint_out will use transport mentioned under this section.

To test outbound calls using the above mentioned Trunk Configuration you may need an internal phone extension. For that, you may configure phone extension 6001 as mentioned below in pjsip.conf itself.

Internal extension / phone configuration

[6001]
type=endpoint
context=Zentrunk
disallow=all
allow=ulaw
auth=6001
aors=6001

[6001]
type=auth
auth_type=userpass
password=password1234
username=6001

[6001]
type=aor
max_contacts=1

You can register any SIP enabled phone with username: 6001 and password: password1234

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Step 2: Dialplan

Next, you should set-up a dialplan. Below mentioned dial plan will dial out to ZENTRUNK using zentrunk_endpoint_out when 6001 (the SIP phone registered with username 6001) dial a number. Note that we have mentioned context=Zentrunk under endpoint 6001.

Add below dial plan in extensions.conf under directory /etc/asterisk.

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extensions.conf

[Zentrunk]
exten => _X.,1,Set(CALLERID(all)="Your Plivo Number" <Your Plivo Number>)
exten => _X.,n,Dial(PJSIP/${EXTEN}@zentrunk_endpoint_out)
exten => _X.,n,Hangup()

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Step 3: Reload Configurations

1. Execute the following command in your terminal to connect to the asterisk CLI:

Asterisk -rvvvv

2. Execute the below command to reload chan_pjsip:

pjsip reload

3. Reload dialplan using below command:

dialplan reload

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Restart Asterisk

Execute the below command from the Linux command line to restart Asterisk.

systemctl restart asterisk