--- schema_version: "1" id: 54fc266b121297b13dd47226007448d9048de75afcae4bdcc0fc156f3d1f0096 title: Configure SIP Trunking in Your Partner Telephony System collection: admin version: current locale: en-us portal_url: https://docs.salesforce.com/admin/current/en-us/ai/264-0-0/telephony_configure_sip_trunking.md canonical_url: https://help.salesforce.com/s/articleView?id=ai.telephony_configure_sip_trunking.htm&release=264.0.0&type=5 source_site: help.salesforce.com content_type: documentation content_hash: 69a97cf88a96095c1f79190125d5faba52c93871c8fc9d6fe2f1d3194fe150e7 description: Before you can configure your telephony system to transfer and escalate calls via SIP, connect your telephony system to Salesforce. In addition to configuring logic to manage traffic and routing, connecting the telephony system to Salesforce also involves creating an encrypted SIP trunk to support user-to-user information exchange and SIP invites. products: - Agentforce - Cross Cloud release: "264" --- # Configure SIP Trunking in Your Partner Telephony System Before you can configure your telephony system to transfer and escalate calls via SIP, connect your telephony system to Salesforce. In addition to configuring logic to manage traffic and routing, connecting the telephony system to Salesforce also involves creating an encrypted SIP trunk to support user-to-user information exchange and SIP invites. Set these attributes for your SIP trunk. - Set the Salesforce SBC endpoint, using a [Salesforce SBC FQDN](https://developer.salesforce.com/docs/ai/agentforce-partner/guide/agentforce-voice-sip-partner-regional-availability.html) and port. - Set the protocol to TLS. - Set the media codecs to audio/PCMU and audio/PCMA. - Set the User to User Information (UUI) header or X-headers before transferring the calls. - Set the UUI encoding format to Hex. - Grant SIP access to IP addresses. Set the allow and deny lists for the transferred calls to Salesforce. ## Sample SIP Trunking Configuration in Genesys To illustrate how to configure SIP trunking in a partner telephony system, this section covers the tasks using Genesys. This example includes UUI header configuration instead of X-headers. For information about configuring a SIP trunk with X-headers, ask your SIP provider for guidance. ### Add a Site Add a site that defines which SIP trunk picks up incoming calls. For the properties not mentioned, keep the default values. 1. Log in to [Genesys Cloud](https://apps.usw2.pure.cloud/directory/). 2. Click the menu and select **Digital and Telephony** | **Telephony** | **Sites**. 3. In the Site page, add a new site. 4. In the Name field, enter a name for the site. For example, enter Salesforce. 5. In the Location field, select **Default PCV Location**. 6. In the Media Regions field, select the region nearest to your site. For example, if your Genesys account is in the US West, select **Americas (US West)**. 7. Save the site. ![The site details in Genesys](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/agentforce_voice_genesys_site.png) ### Create a Number Plan To transfer calls using a SIP trunk, first create a number plan, assign the SIP address to it, and then select the classification for the SIP addresses in the number plan. The number plan determines which SIP trunk to route to. For the properties not mentioned, keep the default values. Later, you’ll configure the SIP trunk. 1. In the Site page, click the Number Plans tab. 2. Click **New Number Plan**. 3. In the Number Plan Name field, enter the name of the plan. 4. In the Match Type field, select an option. For example, select **E.164 Number List**. 5. In the Numbers field, enter the start and end numbers for the range of SIP addresses. If you have only one SIP address, enter the same number for both. Each number must be in the E.164 format. 6. To route incoming calls to the SIP trunk that you will create, in the Classification field, create a classification. To create it, enter the name and click **+**. 7. Save the number plan. ![The number plan for a single-number range](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/agentforce_voice_sip_number_plan.png) ### Create a SIP Trunk Create the SIP trunk. To route incoming calls to the SIP trunk that you created, use the classification that you assigned to the number plan. For the properties not mentioned, keep the default values. 1. In the Digital and Telephony menu, click **Telephony** | **Trunks**. 2. In the External Trunks page, click **Create New**. 3. In the Name field, enter a name for the trunk. 4. In the Trunk Type field, select **Generic BYOC Carrier**. 5. In the Protocol field, select **TLS**. 6. In the Termination Identifier field, enter the termination URI that’s unique within your Genesys Cloud organization's region. Genesys uses this ID to associate inbound calls with the appropriate Genesys Cloud organization’s resources. 7. In the From Site field, choose the site that you created in the previous step. 8. To allow Genesys to call Salesforce, in the SIP Servers and Proxies field, add the Salesforce SBC endpoint. 1. For the SIP Server host name, enter the Salesforce SBC fully qualified domain name (FQDN). For a list of Salesforce SBC FQDNs, see [Agentforce Voice with SIP Regional Availability Chart](https://developer.salesforce.com/docs/ai/agentforce-partner/guide/agentforce-voice-sip-partner-regional-availability.html). Choose the SBC FQDN for your telephony provider and region. 2. For the port, enter 5061. 3. Click **+**. ![The outbound call settings](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/genesys_outbound_properties.png) 9. Under the SIP Access Control section, in the Allow the Following Addresses field, specify the IP address to allow SIP messages from the Salesforce SBC. To get the IP address, perform a lookup against the Salesforce fully qualified domain name. 1. For Windows, open the command prompt and type nslookup SBCFQDN. 2. For Mac or Linux, open the Terminal and type dig SBCFQDN. 10. Under the Identity section, disable **Address Omit + Prefix** for both Calling and Called. 11. Under the Media section, in the Preferred Codec List field, add the supported Salesforce codecs: **audio/PCMU** and **audio/PCMA** and turn on recording for Genesys. 12. Under the Protocol section, enable **Conversation Headers**, enable **UUI Passthrough**, set Type to **User-to-User**, and set Encoding Format to **Hex**. 13. Save the trunk. ![The UUI header settings for a SIP trunk in Genesys.](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/genesys_trunk.png) ### Create an Outbound Route To route calls to the external SIP trunk based on the call’s classification, create an outbound route. The outbound route maps the number plan to a specific trunk. In the site that you created, add an outbound route for the SIP trunk that you created, select the trunk, and select the applicable classification. For the properties not mentioned, keep the default values. 1. In the Outbound Routes tab of the site that you created, click **New Outbound Route**. 2. Enter a name for the outbound route. 3. In the Select External Trunks field, select the trunk that you created. 4. In the Classification field, replace any existing classifications with the one that you created for the number plan. 5. Save the outbound route. ![The outbound route settings for a site in Genesys.](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/genesys_site_outbound_route.png) ### (Optional) Test a Call Optionally, verify that the trunk handles incoming calls correctly. 1. In the Simulate Call tab of the site that you created, enter the SIP address that you added for the number plan. 2. Click **Simulate Call**. Review the status to verify that the call went through successfully. 3. If it fails, review the error message to determine the cause. ![The Simulate Call page shows the status of a simulated call to a SIP address.](https://sf-zdocs-cdn-prod.zoominsoftware.com/tdta-ai-generative_ai-264-0-0-production-enus/f8ef6498-d7b2-44e3-9b34-ffd5a850aae9/generative_ai/images/genesys_site_simulate_call.png)