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 Connect 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.
- 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.
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.
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:
- Navigate to Settings → Account Users.
- Click edit next to the user you wish to update.
- Click or scroll down to the Agent Contact section.
- Use the “Tracking Number” drop-down to select a tracking number to assign to this user.
- 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.
The following is an example username and domain for a SIP user:
SIP Username: a348723u
SIP Domain: http://..............sip.us1.twilio.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 Username, Password).
Alternatively we have a management page for multiple SIP devices that you can use to configure 100s or on request 1000s of phones. The configuration for these can be exported and easily uploaded depending on your VOIP provider.
TO ASSIGN A MULTIPLE SIP DEVICES AT ONCE
- Navigate to Settings → SIP Devices.
- Click Add Phones in the upper right corner.
- Select from the list of Agents to add your VOIP/SIP Devices too.
- Select your CSV file that has phone numbers and agents listed out in multiple rows
- Click upload mapping.
- You may assign a mac address to more easily manage the devices.
- You may also provide details for e911 support enter an address where each phone is physically located.
- Click Create New Devices.
- An export of the list of devices created will be provided that includes the necessary credentials
- You may find it easier to use the export to export to a Polycom or Grand Stream phone system.
Refer to our best practices page to confirm your network is configured properly ad the correct ports are open in your firewall. These ports are applicable to both softphones and SIP phones.