# Connect Dial a SIP URI or phone number. ## **Parameters** | Name | Type | Default | Description | |:|:--|:|:-| | confirmOptional | string | | A URL that returns [SWML](../introduction.mdx) to execute when the call is connected. | | fromOptional | string | Calling partys caller ID number | Caller ID number. Optional. Can be overwritten on each [destination](#parameters-for-destination). | | headersOptional | object | | Custom SIP headers to add to INVITE. Has no effect on calls to phone numbers. | | codecsOptional | string | Based on SignalWire settings | Comma-separated string of codecs to offer. Has no effect on calls to phone numbers. | | webrtc_mediaOptional | boolean | false | If true, WebRTC media is offered to the SIP endpoint. Has no effect on calls to phone numbers. Optional. Default is false. | | session_timeoutOptional | integer | Based on SignalWire settings | Time, in seconds, to set the SIP Session-Expires header in INVITE. Must be a positive, non-zero number. Has no effect on calls to phone numbers. | | ringbackOptional | string[] | Plays audio from the provider | Array of play URIs to play as ringback tone. | | timeoutOptional | integer | 60 seconds | Maximum time, in seconds, to wait for an answer. | | max_durationOptional | integer | 14400 seconds (4 hours) | Maximum duration, in seconds, allowed for the call. | | answer_on_bridgeOptional | boolean | false | Delay answer until the B-leg answers. | | call_state_urlOptional | string | | Webhook url to send call state change notifications to for all legs. Can be overwritten on each [destination](#parameters-for-destination). | | call_state_eventsOptional | string[] | [ended] | An array of call state event names to be notified about. Allowed event names are created, ringing, answered, and ended. Can be overwritten on each [destination](#parameters-for-destination). | In addition, you are required to specify **one and only one** of the following dialing parameters: | Name | Type | Description | |:|:-|:-| | to | string | Single destination to dial. Possible values are a phone number (i.e.: +15552345678) or sip uri (i.e. sip:alice@example.com). | | serial | object[] | Array of [destination](#parameters-for-destination) objects to dial in order. | | parallel | object[] | Array of [destination](#parameters-for-destination) objects to dial simultaneously. | | serial_parallel | object[][] | Array of arrays. Inner arrays contain [destination](#parameters-for-destination) objects to dial simultaneously. Outer array attempts each parallel group in order. | ## **Parameters for destination** | Name | Type | Default | Description | |:|:--|:|:--| | toRequired | string | | Phone number or SIP URI to dial. | | fromOptional | string | Calling partys caller ID number | Caller ID number. Optional. | | timeoutOptional | integer | 60 seconds | Maximum time, in seconds, to wait for destination to answer. | | call_state_urlOptional | string | | Webhook url to send call state change notifications to. | | call_state_eventsOptional | string[] | [ended] | An array of call state event names to be notified about. Allowed event names are created, ringing, answered, and ended. | ## **Variables** Set by the method: - **connect_result:** (out) connected | failed. - **connect_failed_reason:** (out) Detailed reason for failure. - **return_value:** (out) Same value as connect_result. ## **Examples** ### Use connect with Call Fabric {#connect-with-CF} Use a Call Fabric Resource with the connect method by simply including the Resource Address. Learn more by reading our [Introduction to Call Fabric](/guides/administration/guides/signalwire-space/what-is-call-fabric/index.mdx) or the guide to [Managing Resources](/guides/administration/guides/signalwire-space/managing-your-signalwire-resources/index.mdx). yaml andJson sections: main: - answer - play: volume: 10 urls: - silence:1.0 - say:Hello, connecting to a fabric Resource that is a room - connect: to: /public/test_room ### Dial a single phone number yaml andJson version: 1.0.0 sections: main: - connect: from: +15553214321 to: +15551231234 ### Dial a single phone number and go to voicemail on failure yaml andJson version: 1.0.0 sections: main: - connect: from: +15553214321 to: +15551231234 result: case: connected: - hangup default: - execute: voicemail - hangup ### Dial numbers in parallel yaml andJson version: 1.0.0 sections: main: - connect: parallel: - to: +15551231234 - to: +15553214321 ### Dial SIP serially with a timeout yaml andJson version: 1.0.0 sections: main: - connect: timeout: 20 serial: - from: sip:chris@example.com to: sip:alice@example.com - to: sip:bob@example.com codecs: PCMU