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.
Unified session types
The new session type variable, Session Type V2, is now available. This update introduces a range of new fields, variables, and columns that will provide you with access to valuable additional information such as the ability to distinguish between Inbound SMS, Outbound SMS, and Outbound SMS using the API.
You can continue to use your existing scripts and automations while working with your internal teams to plan for the necessary modifications. However, to take advantage of the new fields and variables, you will need to update your scripts, code, automation triggers, and any third-party integrations.
| Area | New field, variable, column names | Legacy field, variable, common names |
|---|---|---|
| API endpoints (/manager/api/v1/calls and /manager/api/v1/chats) | session_type_v2 |
call_type and chat_type will remain |
| Session metadata | session_type_v2 |
|
| {SESSION_TYPE} variable in CRM Record Title in Operation Management | {SESSION_TYPE_V2} | {SESSION_TYPE} will remain |
| CRM tags (Zendesk, Kustomer, Freshdesk) | The new values will be pushed alongside the old values to the same standard tags field | |
| CCAIP Session Object (Salesforce) In order to use the new session type, it is necessary to update to Salesforce Version 1.31 | Session Type v2 | "Session Type" will be renamed to "Session Type v1" and "Channel" will remain |
| Reports | Session Type v2 | "Type" will be renamed to "Type v1" |
| CCAIP Portal | UI in the Portal will contain the new values only | UI in the Portal will contain the new values only |
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 Menu, 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.
Click Save Redirection.