Create routes

Create call routes from the Infobip web interface or by using the Call Routing API.

Route definitions include:

One to 10 destinations, with their respective priority, and optional weight and destination parameters.
(Optional) Inbound traffic filter criteria.

When a route is created, it is assigned a route priority. The first route created is given a priority of 1, and each subsequent route receives a priority that increases by 1. You can change route priorities through the Call Routing user interface or by using call routing APIs.

For more details on filter criteria and route priorities, see Filter-based route selection.

A route can be designated as a Transfer Only Route. This flag designates that the route is only considered for execution when receiving a SIP REFER from an inbound SIP trunk. For more information about SIP REFER, see Reroute calls with SIP REFER.

Destination priorities and weights [#destination-priorities-and-weights-create-routes]

Each entry or destination on a route must be assigned a priority, with values ranging from 1 (highest) to 100 (lowest). You can set priorities in a way that best fits your implementation. When routes are executed, entries are evaluated according to their priority.

Entries in a route that share the same priority must be assigned a weight.

Weight is used to determine the share of the load to each route entry with the same priority.
Priority is used to define the hunt sequence.

Destination specifics [#destinations-specifics-create-routes]

For each destination type in a route, you can define optional elements that produce different results. The following table shows the parameter settings that affect the outcomes.

Destination type	Parameter	Effect
SIP	`username`	When defined, the original destination of the call `to` is overwritten with this value when the call is triggered on the SIP trunk. When not defined, the original destination of the call is preserved when the call is triggered on the SIP trunk.
SIP	`timeOut`	(Optional) Define the time to wait for the call to be established when the call is triggered on the SIP trunk. The `timeOut` may trigger a route advance or go to the next entry in the destination list if it is the first reason to fail the route, next to other internal criteria such as SIP trunk availability.
PHONE	`phoneNumber`	A phone number in the E.164 format. When not specified, this value defaults to the `to` value used in the inbound call.
PHONE	`timeOut`	(Optional) Define the time to wait for the call to be established when this destination is triggered. The `timeOut` may trigger a route advance, for example, go to the next entry in the destination list if it is the first reason to fail the route, next to other internal criteria, such as the destination phone operator's availability.
WEBRTC	`identity`	The identity of the WebRTC user this call must be forwarded to. When not specified, this value defaults to the `to` value used in the inbound call.
WEBRTC	`displayName`	(Optional) Overwrite the name of the WebRTC user to be displayed to the connected WebRTC party.
WEBSOCKET	`endpoint configuration`	A previously defined WebSocket media streaming configuration.
WHATSAPP	`phoneNumber`	A phone number in the E.164 format. When not specified, this value defaults to the `to` value used in the inbound call.
WHATSAPP	`from`	The WhatsApp sender that must be used for this outbound WhatsApp Business Calling call.
VIBER	`phoneNumber`	A phone number in the E.164 format. When not specified, this value defaults to the `to` value used in the inbound call.
VIBER	`from`	The number used as `callerId` for this outbound Viber call. If specified, this must be your Viber Business Call number.
URL	`url`	Your exposed URL must always be specified.
URL	`securityConfig`	(Optional) The security method to authenticate towards your exposed webhook.

Enable recording [#enable-recording-create-routes]

For each entry in a route, you may choose to record the audio of the resulting conversation.

Recordings can be either:

composed (one single audio file with all participants' audio mixed together), or
non-composed (one audio file per participant in the call).

If you want to opt your recordings to your own SFTP server instead of Infobip cloud storage, you can specify an optional text string in filePrefix. The text string is used in the zip file names that are pushed to your server.

Allowed Time Window [#allowed-time-window-create-routes]

Use the Allowed Time Window to control when a route destination is eligible for call routing. This feature gives you precise control over call flows, ensuring that calls are routed based on both the day of the week and specific hour periods. Example use cases for this feature are:

Business Hour Routing: Ensure calls are routed to office numbers only during business hours and redirected to voicemail or alternate destinations after hours.
Special Event Handling: Route calls to special event hotlines only during the event times.
Global Operations: Manage call routing for global offices in different time zones using UTC time to maintain uniformity.

For each destination in a route, administrators can configure one or more Allowed Time Windows. These time windows determine when the destination is valid for routing calls. Multiple time windows can be assigned to a single destination. For example, a destination can be configured to be available:

From Mondays to Fridays, 8 AM to 5 PM.
On Fridays, 8 AM to 6 PM.
On Saturdays, 9 AM to 1 PM.

When multiple time windows are defined for a single destination, the system automatically checks for coherence among these windows to avoid conflicts and overlaps.

When a call triggers a route execution, the system evaluates the current time against the defined Allowed Time Windows for each destination in the route.

If the current time falls outside the defined Allowed Time Windows for a destination, that destination is skipped, and the call is routed to the next eligible destination.

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