Creates a new SipGateway that will be associated with a VoipCarrier

Create a SipGateway

Voxeme API Documentation

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Introduction

Alerts are only kept in the system for seven days.  As there could be a large number of alerts, the api allows you to query pages of data at a time.

Get Alerts

Retrieve the applications that have been created for this account

Get Applications

Create an application, specifying webhook URLs and speech settings.  

Success is indicated by a 201 response containing a json body with a property "sid" that identifies the application_sid that was created.

Create Application

Retrieve a specific application by its unique identifier

Get Application

Update the properties of an application.

Success is indicated by a 204 response.

Update Application

Deletes an application.

Success is indicated by a 204 response

> Note: an application can not be deleted if it is referenced by a phone number or a  carrier.

Delete Application

Retrieve a list of calls for the account that are in progress or recently ended.

> Note: calls that have ended more than ten minutes ago will not be returned.

Get Calls

In this alternative version, the a short form of the webhook is explicitly provided in the payload.

Note that when an application_sid is NOT provided, you must explicitly specify the speech vendor properties in the payload (see example below).

Create call (version 3)

Retrieve information about a specific call in progress, or one that has recently ended (within the past ten minutes).

Get Call

This api allows you to manipulate a call that is in progress -- i.e., "Live Call Control".

The actions available are:
- hangup a call
- mute/unmute a party on a call
- pause/resume the real-time audio stream provided via the [listen](https://www.jambonz.org/docs/webhooks/listen/) command
- transfer one or both parties on a call to new application(s).  Note that you can provide a different application for each party (e.g. queue the called party while the calling party dials a new number)
- play a whisper prompt to a party.
- place or remove a party that is in conference on hold.
- mute/unmute all non-moderators in a conference.

Generally, a single API call can only do one of the above operations.

The JSON payload may contain the properties described in the table below.

| property      | description | when can this be used
| ------------- |-------------| ---------------------|
| call_hook | a new application to start executing on the call for the party associated with the call_sid | at any time a call is active |
| child_call_hook | a new application to start executing on the call for the party associated with the child_call_sid | at any time a call is active and in a [dial](https://www.jambonz.org/docs/webhooks/dial/) verb|
| call_status | Change the status of the call.  Possible values are 'completed' or 'no-answer' (the former terminates an answered call, the latter a call that is ringing) | at any time a call is in-progress or ringing |
| conf_hold_status |Possible values are 'hold' or 'unhold'.  When playing a party on hold, a wait_hook can optionally be provided to play music or a message to the party while on hold.|  Only valid when the party is in a conference. |
| conf_mute_status |Possible values are 'mute' or 'unmute'.  Note that this action applies to all non-morderators in the conference, not to a single member|  Only valid when the party is in a conference. |
| listen_status | Change the status of a listen stream.  Possible values are 'pause' or 'resume'.  Pausing a stream maintains the websocket connection but will discontinue sending audio over the connection.  Resuming will start sending audio again.  This may be useful, for example, when a caller is providing confidential information that you do not want to appear in a recording. | only when a listen command is active on the call (may be nested in an active dial command) | 
| mute_status | Mute or unmute a call that is currently in a Dial verb.  Possible values are 'mute' or 'unmute'. Either party in a call may be modified, depending on the call_sid provided in the path of the request-uri. | only when a dial command is currently active on a call |
| whisper | Play a mid-call whisper prompt to one of the parties on a call in progress.  The whisper prompt is provided in a play or say verb, as shown in the examples below.  The whisper may be played to either party on the call, depending on the call_sid provided in the path of the request-uri. The other party is briefly placed on hold while the prompt is played, and then reconnected to the other party afterwards.| only when a dial command is currently active on the call |
| wait_hook | A relative or absolute url that can be supplied when placing a party that is in conference on hold.| Only when putting a party on conference hold |

The call_status, listen_status, and mute_status properties are mutually exclusive -- only one may be provided in the same request.

The whisper property may be provided alone, or together with a listen_status or mute_status property, in which case the whisper prompt is played after the listen_status or mute_status operation is complete.

Finally, if call_hook is provided, then call_status_hook may also optionally be included.  This is used to specify a new callback to receive call status events.

Success is indicated by a 204 response

Update Call - Live Call Control

Deletes a call.  After this action is done, any updates / Live Call Control operations will not be possible for this call.  

Success is indicated by a 204 response

> Note that deleting a call does not hang up the call, nor does it delete the call detail record for the call.  It merely removes it from the redis database

Delete Call

Call detail records are only kept in the system for 30 days.  As there could be a large number of CDRs, the api allows you to query pages of data at a time.

Get RecentCalls

Retrieve all SpeechCredentials associated with  your account

Get SpeechCredentials

Create SpeechCredential

Get SpeechCredential

Deletes a SpeechCredential.

Success is indicated by a 204 response.

Delete SpeechCredential

Test a SpeechCredential.  This will cause the SpeechCredential to be retrieved and used in a simple text to speech and/or speech recognition request to the speech service.

> Note: the SpeechCredential will only be tested for TTS or STT if it is indicated as being used for that purpose.

API Documentation

v1

Create a SipGateway

Body

Response