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A Session Initiation Protocol Uniform Resource Identifier (SIP URI) destination
refers to a specific address or endpoint that can receive SIP-based
communication. SIP is a protocol used for initiating, modifying, and terminating
real-time communication sessions such as voice and video calls over IP networks.
For more information about session types, see the
session type terminology
documentation.
With as contact list you can create and save a list of SIP URIs,
which are used to identify endpoints in a SIP-based communication network, such
as VoIP (Voice over IP) systems. You can customize call routing and management
in a communication system, depending on the specific needs and requirements of
the organization.
SIP call transfers can be used to route incoming calls to appropriate
destinations based on IVR menu selections or queue routing rules. For example,
a consumer calling your support line may select an option in the IVR menu to be
transferred to a specific department or agent based on their inquiry.
Use cases
Call overflow: SIP call transfers can be used to manage call overflow
situations where a queue becomes too busy or reaches its maximum capacity. Calls
can be automatically transferred to alternative destinations, such as other
queues or backup agents, to ensure efficient call handling and prevent call
abandonment.
Call distribution: SIP call transfers can be used to distribute calls evenly
or according to specific routing rules among agents or departments. This can
help balance the workload and ensure fair distribution of calls, optimizing call
handling efficiency and improving consumer service.
Call consolidation: SIP call transfers can be used to consolidate calls from
multiple sources or channels into a single destination. For example, calls from
different IVR menus or queues can be transferred to a centralized agent or
department for unified handling and streamlined call management.
Configure call transfers at the queue level
You can configure how calls are transferred to a contact in an IVR queue.
To configure call transfers at the queue level, follow these steps:
In the CCAI Platform portal, click Settings > Queue. If you don't
see the Settings menu, click menuMenu, and then click Settings > Queue.
In the IVR (Interactive Voice Response) pane, click Edit / View.
Click the queue that you want to edit.
Go to Automatic Redirection and click the toggle to the show position.
Select Phone number or Outbound SIP transfer.
Do one of the following:
Select Select from contact list. Use the fields that appear to select a
contact list and a destination.
Select Enter phone number. Enter a phone number in the field that
appears.
Select Enter SIP URI address. Enter a SIP URI in the field that
appears. Optionally select Redirect using SIP REFER when available.
[[["Easy to understand","easyToUnderstand","thumb-up"],["Solved my problem","solvedMyProblem","thumb-up"],["Other","otherUp","thumb-up"]],[["Hard to understand","hardToUnderstand","thumb-down"],["Incorrect information or sample code","incorrectInformationOrSampleCode","thumb-down"],["Missing the information/samples I need","missingTheInformationSamplesINeed","thumb-down"],["Other","otherDown","thumb-down"]],["Last updated 2025-08-25 UTC."],[[["\u003cp\u003eSIP URI destinations are specific addresses for receiving SIP-based communications, enabling real-time interactions like voice and video calls over IP networks.\u003c/p\u003e\n"],["\u003cp\u003eContact lists allow users to create and store lists of SIP URIs for identifying endpoints in SIP-based communication networks, facilitating customized call routing.\u003c/p\u003e\n"],["\u003cp\u003eSIP call transfers can route calls to destinations based on IVR selections or queue rules, and they are used for scenarios like call overflow, call distribution, and call consolidation.\u003c/p\u003e\n"],["\u003cp\u003eThe Session Type V2 update introduces new fields, variables, and columns to provide access to additional information and features, such as distinguishing between different types of SMS messages, which requires script updates to be used.\u003c/p\u003e\n"],["\u003cp\u003eCall transfers can be configured at the queue level, allowing for redirection to phone numbers or SIP URI addresses, and it can use a contact list or a manually entered destination.\u003c/p\u003e\n"]]],[],null,["# SIP URIs\n\nA Session Initiation Protocol Uniform Resource Identifier (SIP URI) destination\nrefers to a specific address or endpoint that can receive SIP-based\ncommunication. SIP is a protocol used for initiating, modifying, and terminating\nreal-time communication sessions such as voice and video calls over IP networks.\nFor more information about session types, see the\n[session type terminology](/contact-center/ccai-platform/docs/session-type-terminology)\ndocumentation.\n\nWith as [contact list](/contact-center/ccai-platform/docs/contact-list) you can create and save a list of SIP URIs,\nwhich are used to identify endpoints in a SIP-based communication network, such\nas VoIP (Voice over IP) systems. You can customize call routing and management\nin a communication system, depending on the specific needs and requirements of\nthe organization.\n\nSIP call transfers can be used to route incoming calls to appropriate\ndestinations based on IVR menu selections or queue routing rules. For example,\na consumer calling your support line may select an option in the IVR menu to be\ntransferred to a specific department or agent based on their inquiry.\n\nUse cases\n---------\n\n**Call overflow**: SIP call transfers can be used to manage call overflow\nsituations where a queue becomes too busy or reaches its maximum capacity. Calls\ncan be automatically transferred to alternative destinations, such as other\nqueues or backup agents, to ensure efficient call handling and prevent call\nabandonment.\n\n**Call distribution**: SIP call transfers can be used to distribute calls evenly\nor according to specific routing rules among agents or departments. This can\nhelp balance the workload and ensure fair distribution of calls, optimizing call\nhandling efficiency and improving consumer service.\n\n**Call consolidation**: SIP call transfers can be used to consolidate calls from\nmultiple sources or channels into a single destination. For example, calls from\ndifferent IVR menus or queues can be transferred to a centralized agent or\ndepartment for unified handling and streamlined call management.\n\nConfigure call transfers at the queue level\n-------------------------------------------\n\nYou can configure how calls are transferred to a contact in an IVR queue.\n\nTo configure call transfers at the queue level, follow these steps:\n\n1. In the CCAI Platform portal, click **Settings \\\u003e Queue** . If you don't\n see the **Settings** menu, click menu\n **Menu** , and then click **Settings \\\u003e Queue**.\n\n2. In the **IVR (Interactive Voice Response)** pane, click **Edit / View**.\n\n3. Click the queue that you want to edit.\n\n4. Go to **Automatic Redirection** and click the toggle to the show position.\n\n5. Select **Phone number or Outbound SIP transfer**.\n\n6. Do one of the following:\n\n - Select **Select from contact list**. Use the fields that appear to select a\n contact list and a destination.\n\n - Select **Enter phone number**. Enter a phone number in the field that\n appears.\n\n - Select **Enter SIP URI address** . Enter a SIP URI in the field that\n appears. Optionally select **Redirect using SIP REFER when available**.\n\n7. Click **Save Redirection**."]]
