Adding a SIP-IN line

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SIP (Session Initiation Protocol) technology allows voice calls, and other communication services to be carried over the internet.

You can use SIP-IN for receiving calls from the internet. SIP-OUT, for making outbound calls on the internet, is currently not supported.

For information about how to add a standard Public Switched Telephone Network (PSTN) phone number, see Adding phone numbers.

This article contains the following topics:

• About SIP-IN lines
• Understanding SIP-IN technical requirements
• Understanding SIP-IN line limitations
• Adding a SIP-IN line
• Deleting SIP-IN lines

About SIP-IN lines

When you add a SIP-IN line, you can accomplish the following:

• Call escalation from an AI agent: Connect a third-party AI agent to escalate calls to Zendesk agents when they cannot resolve their query. See Connecting an external AI agent using a SIP-IN line.
• Bring Your Own Carrier (BYOC): Use local carriers and forward calls to Zendesk via SIP to operate their business in that country.
• Call forwarding: Forward calls into Zendesk via SIP-IN without paying PSTN rates for call forwarding.
• Third-party application (external IVR) forwarding: Connect with partner apps during the call flow, such as an external IVR, to enable a more complex and integrated service experience.

Understanding SIP-IN technical requirements

Use the following information to help you understand the technical requirements for SIP-IN lines:

• The AI agent must be able to place a call to a SIP-IN line address, (for example, [e-mail]) configured in Admin Center. Zendesk uses Twilio’s (our service provider's) SIP. Any user can be dialed on the defined SIP URI. The phone number passed in the SIP URI “from” field is used to associate an existing user profile with the ticket.
• Zendesk employs IP access control lists. IP ranges from third parties can be defined in Admin Center. We do not employ any other authentication method such as username and password to authenticate SIP-IN calls.
• SIP-IN lines are compatible with omnichannel routing. When a ticket is generated by the AI agent, it will be directed to agents based on their availability and capacity if you've created omnichannel routing rules.
• Twilio voice media IPs use a single global range;: 168.86.128.0/18 with a UDP port range 10000-60000
• Our provider supports G.711 μ-law (PCMU) and A-law (PCMA) codecs for media.
• SIP-IN lines work with your interactive voice response (IVR) trees. See Routing incoming calls with IVR.
• SIP-IN lines are compatible with customer callback.

[Optional] A ticket ID can be passed to Support to be associated with the SIP call using SIP headers.

• The SIP header format must be: X-Zendesk-Ticket-Id=<ticket number>
• When creating the ticket, use the via_id 34 as detailed [here]. This will identify it as an incoming voice call ticket in Zendesk.

Understanding SIP-IN line limitations

The following limitations currently apply when you create SIP-IN lines:

• Only SIP-IN is available for this release; SIP-OUT is not currently supported for outbound calls.
• The Secure Real-time Transport Protocol (SRTP) is not supported over the SIP. Only the Real-time Transport Protocol (RTP) is supported at this time.
• Failover functionality is not supported on a SIP-IN line.
• SIP-IN lines cannot be configured using APIs. They can only be set up in Admin Center.
• You can’t place outbound calls from a SIP-IN line. In the case of call overflow and agent forwarding, you’ll need to choose an external PSTN number for outbound calls.
• You can’t block numbers from a SIP-IN line.

Adding a SIP-IN line

Talk admins can create SIP-IN lines.

To add a SIP-IN line

1. In Admin Center, click Channels in the sidebar, then select Talk and email > Talk.
2. Click the Lines tab, where you can review your existing phone numbers, digital lines, and SIP lines and add new ones.
3. Click Add Line > Add SIP line.

   

4. On the SIP configuration tab, configure the following settings:
   • Nickname: Enter a unique name for the SIP line. You'll need this to identify the line in reports, integrate it with your apps, and let your agents know the SIP line they are being called on.
   • URI: Enter the SIP URI. This address is the URI of your SIP-IN line created in Zendesk, for example [e-mail]. You use this URI to send calls to Zendesk.
   • Authentication: Click Add IP address range to add a range of allowed IP addresses in the form of a CIDR (Classless Inter-Domain Routing) address. Authentication is based on the source IP address of the SIP request.
5. On each tab of the line settings page, configure the settings you want using the following articles as a reference:
   • Settings tab: See Managing line settings.
   • Voicemail tab: See Configuring voicemail options.
   • Routing tab: See Routing incoming calls to groups of agents.
   • Call recording tab: See Managing call recording options.
   Note: Not all of the settings will be available for SIP lines.
6. When you are finished, click Save.

Your SIP-IN line is now set up.

Editing SIP-IN lines

After you’ve created a SIP line, you can edit its settings if you want to change anything.

To edit a SIP line

1. In Admin Center, click Channels in the sidebar, then select Talk and email > Talk.
2. From the list of SIP lines, click the gear icon () next to the SIP-IN line you want to edit, and then select Edit.

   

3. Edit the settings as required using the instructions in the Adding a SIP-IN line section.
4. Click Save changes.

Deleting SIP-IN lines

After you’ve created a SIP line, you can delete it if you no longer need it.

To delete a SIP line

1. In Admin Center, click Channels in the sidebar, then select Talk and email > Talk.
2. From the list of SIP lines, click the gear icon () next to the SIP line you want to delete, and then select Delete.
3. Click Delete SIP line to confirm you want to delete the SIP line.