SIP Caller with FreePBX

SIP Caller with FreePBX

Overview

Create an extension in FreePBX

Obtain credentials from FreePBX

Optional - Install the SIP Caller Proxy for Asterisk

Overview

SIP Caller connects to your Phone System as a remote extension. Therefore, in order to use SIP Caller with FreePBX you will need to create an extension for it, and then obtain the SIP credentials.

Create an extension in FreePBX

In order to create an extension in FreePBX proceed as follows:

1. In the management console, go to “Applications” > “Extensions”.

2. Select the menu “Add Extension” > “Add New SIP [chan_pjsip] Extension”.

3. Enter the extension number in the “User Extension” field, and assign a display name, and optionally an outbound caller ID.
4. You will need the values from “User Extension” and “Secret”, so copy them and keep them handy when you configure the extension in SIP Caller.
5. In the “Advanced” tab, search for the option “Outbound Concurrency Limit” and set the value to the number of simultaneous calls you want SIP Caller to make.
6. Click the “Apply config” button on the upper right corner, so the PBX starts working with this extension.

Obtain credentials from FreePBX

SIP Caller will require the “User Extension” and “Secret” values obtained from the previous step. When you configure the extension in SIP Caller, you will need to use those values as follows:

• Username: User Extension
• Auth ID: User Extension
• Password: Secret

For detailed instructions on how to configure the Phone System in SIP Caller, check here.

Optional - Install the SIP Caller Proxy for Asterisk

SIP Caller supports running campaigns in “Predictive Dialer” mode. This means that the number of calls made will be adjusted in real time, considering the number of agents available in the queue to handle these calls. In order to do this, SIP Caller needs to know the status of the queue, i.e. the number of waiting calls, and the number of available and busy agents. This information is reported to SIP Caller by the “SIP Caller Proxy” module that must be installed in the PBX server. This is not required to run campaigns in “Power Dialer” mode.

In order to install the “SIP Caller Proxy” for Asterisk, proceed as follows:

1. The “SIP Caller Proxy” module connects to your FreePBX / Asterisk server using the Asterisk Manager Interface (AMI). Therefore, you will need to create a user for SIP Caller to connect through this protocol. When using FreePBX:

• Go to Settings > Asterisk Manager Users, and click the “Add Manager” button.

• Enter a name, for example “sipcaller”, and configure the network restrictions as needed (the module will only connect locally).

• In the Permissions tab, SIP Caller only requires that you enable “System” > “Read” permission, as it just executes the “QueueSummary” AMI action.

2. When using a different asterisk flavor, you will need to edit the “/etc/asterisk/manager.conf” configuration file, and add a user for SIP Caller there:

[sipcaller]

secret=a_strong_password_here

deny=0.0.0.0/0.0.0.0

permit=127.0.0.1/255.255.255.0

read=system

3. In the SIP Caller management console, go to Settings > API Keys, and create an API Key assigning the “Phone System Proxy” role. This API Key will be used by the “SIP Caller Proxy” module to connect to the SIP Caller REST API and report the status of the queues in real time.
4. Go to Phone Systems > Edit your Phone System > Queue Monitoring tab, and follow the instructions to install the “SIP Caller Proxy” module in your Asterisk server. You will need SSH access with a user who can execute commands as root via `sudo`.
5. When running the installation script in your FreePBX/Asterisk server, you will need to provide the following information:

• The API Key Token for the API Key created above.
• The port number on which the Asterisk Management Interface (AMI) is listening for requests (by default 5038).
• The user name to use to connect to AMI (this is the user created in the previous step).
• The password assigned to this user.