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SIP Setup

Support Hub Flows Routing › SIP Setup
Last Updated: 04/24/2019

The CallTrackingMetrics platform can be enabled to allow routing to SIP devices.  However, due to the nature of SIP routing and its dependency on your network and device, we cannot provide support or troubleshooting for SIP systems outside of the SIP settings in CallTrackingMetrics.

SIP can only be enabled for Contact Center accounts.

Before beginning a SIP setup, please read this article carefully.  We strongly recommend considering the following:

  • The built-in CTM softphone is an easy-to-use alternative that requires no setup or advanced configuration.  Click here to read more about using the softphone.
  • You will need a network administrator or a third party network expert who can help you configure your network and your SIP device.
  • The majority of problems with SIP/VoIP routing are caused by network issues.  You will need to have a strong network with QoS.  Run this bandwidth test before configuring your SIP system to get a general idea of how many simultaneous phone connections your network can support.


Enabling SIP in CallTrackingMetrics

To enable SIP for your CTM account, navigate to Settings → Account Settings and click or scroll down to Behaviors.  Find the toggle labeled “Allow SIP Devices for inbound & outbound dialing” and click to turn the toggle to ON.  Carefully review the pop-up that appears.  Click “I Understand” to agree to the terms and enable SIP.


SIP Devices

You will need to acquire SIP phones for each agent or workstation that you would like to route calls to.  Importantly, some networks that provide SIP phones have those phones pre-configured to use their own software.  If you are moving over from a previous SIP setup and wish to keep your devices, you may need to restore factory default settings before the phone can be used on a network other than the one it was originally intended for.

While CTM does not officially endorse a particular device at this time, customers have reported good experiences using the Grandstream GXP1625 with our system.  You can find documentation for these phones here:


CTM User Profiles and Setup

Each agent that will be using a SIP device will need a user login for your CTM account.  Click here to learn how to create a user in your account.

Once SIP is enabled in your account, user profiles will have now have a SIP/VoIP Phone section in their user profile page.  Each user needs to have an assigned tracking number in the Agent Contact section of the user profile page.

To assign a tracking number to a user:
  1. Navigate to Settings → Account Users.
  2. Click edit next to the user you wish to update.
  3. Click or scroll down to the Agent Contact section.
  4. Use the “Tracking Number” drop-down to select a tracking number to assign to this user.
  5. Click Save Changes.

Our system will generate a SIP Hostname and a SIP Username for the CTM user.  Create a password for the SIP connection and enter it into the Change Password field in the SIP section of the CTM user profile.  Combine the SIP Username and SIP Hostname to identify the SIP URI (see example below) in the following format:

sip:[SIP Username]@[SIP Hostname]

The initial “sip:” indicates the connection protocol between systems/devices (ie: use SIP ports/connection protocol).

SIP Settings


The following is an example username, hostname, and URI for a SIP user:

SIP Username: agent12345432123454321
SIP Hostname: phone.plivo.com
SIP URI: sip:agent12345432123454321@phone.plivo.com

You will then need to add a new connection to the agent’s SIP device/phone.  Use the information from the CTM user profile to create that connection (SIP URI, SIP Username, Password).


Client Network

Refer to our best practices page to confirm your network is configured properly.

You will need to make sure the following ports (port ranges) are open in your firewall.  These ports are applicable to both softphones and SIP phones.

  • UDP Port Range 5060 through 5065
  • UDP Port Range 10,000 through 30,000
  • RTP Port Range 10,000 through 40,000
  • SIP phones specifically need TCP and UDP ports 5060 and 5061 open.

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