sipgateio
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The official Node.js library for sipgate.io
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sipgateio-node
A JavaScript library for sipgate.io
Get Started
Installation
For use in node applications you can install with
npm install sipgateio --save
Alternatively, a bundled version can be obtained from the github releases page.
Or use a CDN like jsDelivr instead:
https://cdn.jsdelivr.net/gh/sipgate-io/sipgateio-node@latest/bundle/sipgate-io.min.js
Available Functionality
The following features are already implemented in the current version of this library:
SMS
Send text messages, either instantly or scheduled. The caller ID can be set from the sipgate web interface, the default is the string "sipgate".
Fax
Send any PDF file as a fax to a single number.
Call
Initiate a call between two phones of your choice – no matter if inside your sipgate account or outside.
Webhook (node.js only)
Set up a webserver to process real-time call data from sipgate.io.
Webhook Settings
Configure the webhook functionality of sipgate.io. Currently, you can set URLs and whitelist extensions for triggering webhooks as well as toggle the debug log.
Fluent interface
The fluent interface module is a wrapper for the webhook module to simplify its usage.
Contacts
Import contacts in CSV format into your sipgate account.
History
Fetch multiple or a specific event from your history.
Numbers
Receive all numbers assigned to your account.
Real Time Call Manipulation (RTCM)
Manipulate present calls that containts playing announcements with own audio-files. Also you can hangup, hold, mute an exist call and send DTMF singals.
Usage
Creating a Client
You can connect the client by passing a valid OAuth token (You have to implement the OAuth flow yourself). You can find more information on our API documentation.:
const client = sipgateIO({
token: '<valid Token>',
});
As alternative you can use personal access tokens to authenticate. You can find more information on our API documentation.
const client = sipgateIO({
tokenId: '<your tokenId>',
token: '<your personal access token>',
});
Possible authentication objects:
interface OAuthCredentials {
token: string;
}
interface PersonalAccessTokenCredentials {
tokenId: string;
token: string;
}
The sipgateIO
method accepts your valid sipgate credentials as defined in the AuthCredentials
type and returns a sipgate.io Client.
The client contains as members the supported modules (sms
, fax
, call
, etc.).
The client provides the following functions:
async function getAuthenticatedWebuserId(): Promise<string>;
async function getWebUsers(): Promise<Webuser[]>;
The getAuthenticatedWebuserId()
method returns the ID of the admin account.
The getWebUsers()
method returns an array of the following type:
export interface Webuser {
id: string;
firstname: string;
lastname: string;
email: string;
defaultDevice: string;
busyOnBusy: string;
addressId: string;
directDialIds: string[];
timezone: string;
admin: string;
}
Note that creating the sipgate.io Client will not validate your credentials, you might want to call the method client.getAuthenticatedWebuserId()
to check that your credentials are correct.
SMS
The SMS module provides the following functions:
async function send(sms: ShortMessage): Promise<void>;
async function schedule(sms: ShortMessage, sendAt: Date): Promise<void>;
async function getSmsExtensions(webuserId: string): Promise<SmsExtension[]>;
Note: sendAt
can be 30 days in advance at max.
Note: you should be aware that the request will take a short time to be processed. Values for sendAt
should not just be a few seconds in the future. If sendAt is in the past an error will be thrown.
The ShortMessage
type requires the following fields:
interface ShortMessage {
smsId: string;
to: string;
message: string;
}
or
interface ShortMessage {
from: string;
to: string;
message: string;
}
The getSmsExtensions
method returns an array of the following type:
export interface SmsExtension {
id: string;
alias: string;
callerId: string;
}
Fax
The fax module provides the following functions:
async function send(fax: Fax): Promise<SendFaxSessionResponse>;
async function getFaxStatus(sessionId: string): Promise<FaxStatus>;
async function getFaxlines(): Promise<Faxline[]>;
async function getFaxlinesByWebUser(webuserId: string): Promise<Faxline[]>;
The send
function allows you to send a fax by passing an object with the following fields:
interface Fax {
to: string;
fileContent: Buffer;
filename?: string;
faxlineId?: string;
}
and returns an SendFaxSessionResponse
:
interface SendFaxSessionResponse {
sessionId: string;
}
By using getFaxStatus
and passing the sessionId
you received from the send
function, you will receive one of the following values:
enum FaxStatus {
SENT = 'SENT',
PENDING = 'PENDING',
FAILED = 'FAILED',
SENDING = 'SENDING',
SCHEDULED = 'SCHEDULED',
}
With the getFaxlines
function you can get a list of faxlines that belong to the currently authenticated user.
It returns an array of the following type:
export interface Faxline {
id: string;
alias: string;
tagline: string;
canSend: boolean;
canReceive: boolean;
}
With the getFaxlinesByWebUser
function you can get a list of faxlines that belong to the specified user. It also returns an array of Faxline
.
Call
The call module provides the following function:
async function initiate(
newCallRequest: CallData
): Promise<InitiateNewCallSessionResponse>;
The CallData
contains the following fields:
interface CallData {
deviceId?: string;
from: string;
to: string;
callerId?: string;
}
The InitiateNewCallSessionResponse
contains only a session ID:
interface InitiateNewCallSessionResponse {
sessionId: string;
}
CallData details
The following table shows valid parameter combinations
| to | from | callerId | deviceId | | ------ | --------- | -------- | --------- | | number | extension | - | - | | number | extension | number | - | | number | extension | number | extension | | number | number | - | extension | | number | extension | - | extension | | number | number | number | extension |
The displayed number at the callee device is determined by a hierarchy:
If not set it falls back to the next stage:
callerId
deviceId
(related phone number)from
(related phone number iffrom
is an extension)
The param deviceId
is only mandatory if from
is not an extension.
Valid extension types are e, p, x and y.
| extension type | phone type | | :------------: | -------------- | | e | VoIP phone | | p | user phoneline | | x | external phone | | y | mobile phone |
Scenario 1: basic call
const call = createCallModule(client);
const callData = {
from: 'e14',
to: '+4921165432',
};
call.initiate(callData);
Behavior:
The phone with extension e14
rings first, after pick-up the callee
is called. The default number associated with e14
is displayed at the callee device.
Scenario 2: custom caller id
const callData = {
from: 'p0',
to: '+4921165432',
callerId: '+4917012345678',
};
call.initiate(callData);
Behavior:
Same situation as previous example, but displayed number is now callerId
(see hierarchy).
Scenario 3: group call
const callData = {
from: '+4921123456',
deviceId: 'e14',
to: '+4921165432',
};
call.initiate(callData);
If the from
number refers to a group of phones rather than a single one all phones in the group will ring and the first to be picked up will establish the call with the to
.
The deviceId
is needed for billing and determines the number which will be displayed at the to
device. For instance, e14
has the default number '+4921156789'.
Webhooks
Please note: The webhook feature is only available in node.js environments.
What is a webhook?
The Webhook API provides processing of real-time call data.
A webhook is a POST request that sipgate.io makes to a predefined URL when a certain event occurs. These requests contain information about the event that occurred in application/x-www-form-urlencoded format.
The webhook module provides a simple means to set up a server for handling these webhooks.
The following types of events can trigger webhooks:
| EventType | Description | | --------- | --------------------------------------- | | NEW_CALL | signals that a new call is ringing | | ANSWER | signals that the call has been answered | | HANGUP | signals that the call has been hung up | | DATA | signals dtmf tones sent in the call |
For any of those events, a callback function can be registered to be called upon receiving the respective webhook.
Additionally, for the types NEW_CALL
and DATA
a response may be returned containing commands to trigger actions like hanging up or redirecting calls.
For generating that response our library provides a convenient response interface.
Usage
To begin, instantiate the webhook module by calling createWebhookModule
. The resulting object provides only one method, createServer
which takes a configuration object of type ServerOptions
.
export interface ServerOptions {
// the local port to listen on
port: number;
// the publicly accessible server address
// (including the port, if not standard)
serverAddress: string;
// an optional hostname (default: localhost)
hostname?: string;
// you can optionally skip the signature verification
skipSignatureVerification?: boolean;
}
It returns a Promise<WebhookServer>
which, when resolved, provides the following methods:
interface WebhookServer {
onNewCall: (fn: HandlerCallback<NewCallEvent, ResponseObject | void>) => void;
onAnswer: (fn: HandlerCallback<AnswerEvent, void>) => void;
onHangup: (fn: HandlerCallback<HangUpEvent, void>) => void;
onData: (fn: HandlerCallback<DataEvent, ResponseObject | void>) => void;
stop: () => void;
}
The stop
method simply kills the server, the other methods each take a callback function for handling the respective types of events suggested by their name.
Security
Webhooks are verified to ensure that they are sent by sipgate.
You can optionally disable this via the skipSignatureVerification
flag.
Additionaly you can check that webhooks are originated from push-api.sipgate.net (217.116.118.254 or 212.9.46.32).
Registering event callbacks
Each of the four callback registration methods takes a single callback function that accepts a webhook object of the respective type (i.e. NewCallEvent
, AnswerEvent
, HangUpEvent
, or DataEvent
). In the case of onNewCall
and onData
the provided function may return a ResponseObject
(details below)
Within the callback function the following fields are accessible:
All event types
NewCall
The NewCallEvent
type offers the following fields:
| Parameter | Description |
| ---------------- | ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| event
| EventType.NEW_CALL
|
| from
| The calling number (e.g. "492111234567"
or "anonymous"
) |
| to
| The called number (e.g. "4915791234567"
) |
| direction
| The direction of the call (either "in"
or "out"
) |
| callId
| A unique alphanumeric identifier to match events to specific calls. |
| originalCallId
| A unique alphanumeric identifier to match events to specific calls across transfers. |
| user
| The sipgate user(s) involved. It is the name of the calling user when direction is "out"
, or of the users receiving the call when direction is "in"
. Group calls may be received by multiple users. In that case a user
parameter is set for each of these users. |
| userId
| The IDs of sipgate user(s) involved (e.g. "w0"
). |
| fullUserId
| The full IDs of sipgate user(s) involved (e.g. "1234567w0"
). |
| xcid
| Another unique alphanumeric identifier to match events to specific calls |
Answer
the AnswerEvent
type offers the following fields:
| Parameter | Description |
| ----------------- | ----------------------------------------------------------------------------------------------- |
| event
| EventType.ANSWER
|
| from
| The calling number (e.g. "492111234567"
or "anonymous"
) |
| to
| The called number (e.g. "4915791234567"
) |
| direction
| The direction of the call (either "in"
or "out"
) |
| callId
| A unique alphanumeric identifier to match events to specific calls. |
| originalCallId
| A unique alphanumeric identifier to match events to specific calls across transfers. |
| user
| Name of the user who answered this call. Only incoming calls can have this parameter |
| userId
| The IDs of sipgate user(s) involved (e.g. "w0"
). |
| fullUserId
| The full IDs of sipgate user(s) involved (e.g. "1234567w0"
). |
| answeringNumber
| The number of the answering destination. Useful when redirecting to multiple destinations |
| diversion
| If a call was diverted before it reached sipgate.io this contains the originally dialed number. |
| xcid
| Another unique alphanumeric identifier to match events to specific calls |
HangUp
the HangUpEvent
type offers the following fields:
| Parameter | Description |
| ----------------- | ----------------------------------------------------------------------------------------------- |
| event
| EventType.HANGUP
|
| from
| The calling number (e.g. "492111234567"
or "anonymous"
) |
| to
| The called number (e.g. "4915791234567"
) |
| direction
| The direction of the call (either "in"
or "out"
) |
| callId
| Same as in NewCallEvent
for a specific call |
| originalCallId
| A unique alphanumeric identifier to match events to specific calls across transfers. |
| cause
| The cause for the hangup event (see table below) |
| answeringNumber
| The number of the answering destination. Useful when redirecting to multiple destinations |
| diversion
| If a call was diverted before it reached sipgate.io this contains the originally dialed number. |
| xcid
| Another unique alphanumeric identifier to match events to specific calls |
Hangup causes
Hangups can occur due to these causes:
| Cause | Description |
| ------------------ | --------------------------------------------------------------- |
| "normalClearing"
| One of the participants hung up after the call was established |
| "busy"
| The called party was busy |
| "cancel"
| The caller hung up before the called party picked up |
| "noAnswer"
| The called party rejected the call (e.g. through a DND setting) |
| "congestion"
| The called party could not be reached |
| "notFound"
| The called number does not exist or called party is offline |
| "forwarded"
| The call was forwarded to a different party |
Data
the DataEvent
type offers the following fields:
| Parameter | Description |
| ---------------- | ---------------------------------------------------------------------------------------- |
| event
| EventType.DATA
|
| callId
| Same as in NewCallEvent
for a specific call |
| originalCallId
| A unique alphanumeric identifier to match events to specific calls across transfers. |
| dtmf
| Digit(s) the user has entered. If no input is received, the value of dtmf will be empty. |
Sending a response
For composing an XML response from withing a callback function our library offers a convenient response interface:
interface WebhookResponseInterface {
redirectCall: (redirectOptions: RedirectOptions) => RedirectObject;
sendToVoicemail: () => VoicemailObject;
rejectCall: (rejectOptions: RejectOptions) => RejectObject;
playAudio: (playOptions: PlayOptions) => PlayObject;
playAudioAndTransfer: (
playOptions: PlayOptions,
transferOptions: TransferOptions,
client: SipgateIOClient,
callId: string,
timeout?: number
) => Promise<PlayObject>;
playAudioAndHangUp: (
playOptions: PlayOptions,
client: SipgateIOClient,
callId: string,
timeout?: number
) => Promise<PlayObject>;
gatherDTMF: (gatherOptions: GatherOptions) => GatherObject;
hangUpCall: () => HangUpObject;
}
Redirecting calls
The redirectCall
method accepts an options object of type RedirectOptions
with the following fields:
type RedirectOptions = {
numbers: string[];
anonymous?: boolean;
callerId?: string;
};
Sending calls to voicemail
The sendToVoicemail
method accepts no further options.
Rejecting calls
The rejectCall
method accepts an options object of type RejectOptions
with a single field, the reason for rejecting the call. This reason may be one of the following:
enum RejectReason {
BUSY = 'busy',
REJECTED = 'rejected',
}
Play audio
The playAudio
method accepts an options object of type PlayOptions
with a single field, the URL to a sound file to be played.
Note: Currently the sound file needs to be a mono 16bit PCM WAV file with a sampling rate of 8kHz. You can use conversion tools like the open source audio editor Audacity to convert any sound file to the correct format.
Linux users might want to use mpg123 to convert the file:
mpg123 --rate 8000 --mono -w output.wav input.mp3
Play audio and hang up
The playAudioAndHangUp
method accepts an options object of type PlayOptions
with a single field, the URL to a sound file to be played.
In addition, this also requires a sipgateIOClient
, a unique callId
from an current active call and a timeout
which is optional.
After the audio file has been played and the additional timeout has expired, the call is terminated based on the callId
.
Note: For any information about the audio file please look at play audio.
Play audio and transfer
The playAudioAndTransfer
method accepts an options object of type PlayOptions
with a single field, the URL to a sound file to be played, and an object of type TransferOptions
, which contains the phone number to which the call should be transferred.
In addition, this also requires a sipgateIOClient
, a unique callId
from an current active call and a timeout
which is optional.
After the audio file has been played and the additional timeout has expired, the call is terminated based on the callId
.
Note: For any information about the audio file please look at play audio.
Gather DTMF tones
The gatherDTMF
method accepts an options object of type GatherOptions
with the following fields:
type GatherOptions = {
announcement?: string;
maxDigits: number;
timeout: number;
};
maxDigits
(> = 1) specifies to maximum number of DTMF tones to be gathered , the timeout
(> = 0) is the period in milliseconds to wait for DTMF input from a caller before processing. Please note that the establishment of the call is delayed until this period has elapsed.
By specifying a URL to a sound file as announcement
an audio message can be played to inform callers what DTMF tones they should send.
Note: Please consider the above restrictions concerning the format of the announcement file.
Hang up calls
The hangUpCall
method accepts no further options.
Webhook Settings
The webhook settings module provides the following functions to update settings:
async function setIncomingUrl(url: string): Promise<void>;
async function clearIncomingUrl(): Promise<void>;
These two functions allow for the setting and clearing of the URL to be called when a webhook is triggered by an incoming call.
async function setOutgoingUrl(url: string): Promise<void>;
async function clearOutgoingUrl(): Promise<void>;
Analogous functions exist for the URL that handles outgoing calls.
async function setWhitelist(extensions: string[]): Promise<void>;
async function clearWhitelist(): Promise<void>;
async function disableWhitelist(): Promise<void>;
The whitelist specifies extensions that should trigger webhooks.
By default, webhooks are enabled for all phoneline and group extensions.
This behavior is restored by calling disableWhitelist
.
The disableWhitelist
completely removes the whitelisting and enables all phoneline and group extensions.
async function setLog(value: boolean): Promise<void>;
The setLog
function toggles, the function to display all incoming and outgoing events, which have been sent to your Incoming
and Outgoing
Url.
These parameters can be set using these functions: setIncomingUrl
and setOutgoingUrl
.
interface WebhookSettings {
incomingUrl: string;
outgoingUrl: string;
log: boolean;
whitelist: string[] | null;
}
async function getWebhookSettings(): Promise<WebhookSettings>;
The getWebhookSettings
function returns you the current settings of sipgate.io including the incoming/outgoing URL, whether logging is enabled or if a specific whitelist of devices is set for your incoming & outgoing URL.
Fluent interface
The fluent interface module currently provides a wrapper for the following functions of the webhook module:
- port
- serverAddress
- AnswerCallback
- DataCallback
- HangUpCallback
- NewCallCallback
- startServer
Example:
new FluentWebhookServer()
.setServerPort(port)
.setServerAddress(serverAddress)
.setOnNewCallListener((newCallEvent) => {
console.log(`New call from ${newCallEvent.from} to ${newCallEvent.to}`);
})
.setOnAnswerListener((answerEvent) => {
console.log(`Answer from: ${answerEvent.from}`);
})
.setOnHangupListener((hangupEvent) => {
console.log(`Hangup with cause: ${hangupEvent.cause}`);
})
.setOnDataListener((dataEvent) => {
console.log(`Data from Call: ${dataEvent.originalCallId}`);
})
.startServer();
Contacts
The contacts module provides the following functions:
interface ContactImport {
firstname: string;
lastname: string;
address?: Address;
phone?: PhoneNumber;
email?: Email;
organization?: string[];
}
type ContactUpdate = ContactResponse;
interface ContactResponse {
id: string;
name: string;
picture: string;
emails: Email[];
numbers: PhoneNumber[];
addresses: Address[];
organization: string[][];
scope: Scope;
}
interface ContactsModule {
create: (contact: ContactImport, scope: Scope) => Promise<void>;
update: (contact: ContactUpdate) => Promise<void>;
delete: (id: string) => Promise<void>;
deleteAllPrivate: () => Promise<void>;
deleteAllShared: () => Promise<void>;
importFromCsvString: (csvContent: string) => Promise<void>;
importVCardString: (vcardContent: string, scope: Scope) => Promise<void>;
exportAsCsv: (
scope: ExportScope,
delimiter?: string,
pagination?: Pagination,
filter?: ContactsExportFilter
) => Promise<string>;
exportAsVCards: (
scope: ExportScope,
pagination?: Pagination,
filter?: ContactsExportFilter
) => Promise<string[]>;
exportAsSingleVCard: (
scope: ExportScope,
pagination?: Pagination,
filter?: ContactsExportFilter
) => Promise<string>;
get: (
scope: ExportScope,
pagination?: Pagination,
filter?: ContactsExportFilter
) => Promise<ContactResponse[]>;
}
The create
method:
Given a ContactImport
and Scope
, creates a contact.
The update
method:
Takes a ContactResponse
, the output of ContactsModule.get
, and updates the corresponding contact.
The delete
method:
Deletes a contact by a given id.
The deleteAllPrivate
method:
Deletes all private contacts.
The deleteAllShared
method:
Deletes all shared contacts.
The import
method:
It takes a valid ContactImport
Object and creates a Contact in the requested Scope
.
The importFromCsvString
method:
It takes a valid CSV-formatted string (columns separated by ",") containing at least the following fields:
- firstname
- lastname
- number
These fields may be provided in an arbitrary order. Additional fields as well as empty lines will be ignored. Empty records (i.e. just separators) produce a warning but no error. The same is true for strings containing only the header row.
Example:
lastname,firstname,number
Turing,Alan,+4921163553355
Lovelace,Ada,+4921163553355
The importVCardString
method:
It takes a valid VCard 4.0 string, containing at least the following fields:
name
containsfirstname
andlastname
number
The exportAsCsv
method:
It returns a csv strings containing all contacts for the given scope.
You can also add a specific delimiter for the csv format.
Note: using a filter will ignore pagination
The exportAsJSON
method:
It returns JSON objects containing all contacts for the given scope.
Note: using a filter will ignore pagination
The exportAsVCards
method:
It returns mulitple vCard-strings containing all contacts for the given scope
Note: using a filter will ignore pagination
The exportAsSingleVCard
method:
It returns a vCard-address-book containing all contacts for the given scope
Note: using a filter will ignore pagination
The exportAsObjects
method:
It returns a list of contacts for the given scope as described in the following interface.
Note: using a filter will ignore pagination
interface ContactRequest {
id: string;
name: string;
picture: string;
emails: { email: string; type: string[] }[];
numbers: { number: string; type: string[] }[];
addresses: Address[];
organization: string[][];
scope: Scope;
}
Scopes
The PRIVATE
Scope contains all contacts created by yourself and not shared with other people.
The SHARED
Scope includes all contacts created by anyone in your organization and are therefore shared with other people.
The INTERNAL
Scope contains the contacts which are created by sipgate
such as a contact for any webuser
in your organization.
Adress and Numbers:
You can only save one address and one number using the Format.
History
The history module provides functionality to fetch all or specific history events.
interface HistoryModule {
fetchAll: (
filter?: HistoryFilter,
pagination?: Pagination
) => Promise<HistoryEntry[]>;
fetchById: (entryId: string) => Promise<HistoryEntry>;
deleteByListOfIds: (entryIds: string[]) => Promise<void>;
deleteById: (entryId: string) => Promise<void>;
batchUpdateEvents: (
events: HistoryEntry[],
callback: (entry: HistoryEntry) => HistoryEntryUpdateOptions
) => Promise<void>;
exportAsCsvString: (
filter?: BaseHistoryFilter,
pagination?: Pagination
) => Promise<string>;
}
The fetchAll
method:
The fetchAll method can filter the result by using the 'HistoryFilter' interface. You can decide how many history events you receive by adjusting the values in the pagination object.
interface HistoryFilter {
connectionIds?: string[];
types?: HistoryEntryType[];
directions?: Direction[];
archived?: boolean;
starred?: boolean;
from?: Date;
to?: Date;
phonenumber?: string;
}
interface Pagination {
offset?: number;
limit?: number;
}
fetchById
and fetchAll
methods returns one or multiple history events described by the following base-structure:
interface BaseHistoryEntry {
id: string;
source: string;
target: string;
sourceAlias: string;
targetAlias: string;
type: HistoryEntryType;
created: Date;
lastModified: Date;
direction: Direction;
incoming: boolean;
status: string;
connectionIds: string[];
read: boolean;
archived: boolean;
note: string;
endpoints: RoutedEndpoint[];
starred: boolean;
labels: string[];
}
There are multiple event-types, such as:CallHistoryEntry
, FaxHistoryEntry
, SmsHistoryEntry
, VoicemailHistoryEntry
.
A more detailed description of these types can be found here.
The batchUpdateEvents
function
You have to pass events, which should be updated, to the batchUpdateEvents
function. The second parameter is a mapping function which gives you the ability to return HistoryEntryUpdateOptions
for every event which is used to set the specified parameters in the history entry.
export interface HistoryEntryUpdateOptions {
archived?: boolean;
starred?: boolean;
note?: string;
read?: boolean;
}
History deletion
The deleteById
method allows you to delete an history entry with the given id.
The deleteByListOfIds
method allows you to delete multiple history entries by a given list of ids.
The exportAsCsvString
method
The exportAsCsvString
method allows you to export your history entries as a csv string.
Optionally you can filter and paginate the response by using the following parameters:
export interface BaseHistoryFilter {
connectionIds?: string[];
types?: HistoryEntryType[];
directions?: HistoryDirection[];
archived?: boolean;
starred?: boolean;
startDate?: Date;
endDate?: Date;
}
interface Pagination {
offset?: number;
limit?: number;
}
Numbers
The numbers module provides the following function:
interface NumbersModule {
getAllNumbers: (pagination?: Pagination) => Promise<NumberResponseItem[]>;
}
The getAllNumbers
method:
The method returns an array of objects with the type NumberResponseItem
, which has the following structure:
interface NumberResponseItem {
id: string;
number: string;
localized: string;
type: NumberResponseItemType;
endpointId: string;
endpointAlias: string;
endpointUrl: string;
mnpState?: NumberMnpState;
portId?: number;
}
Where NumberResponseItemType
is one value out of the following enum:
enum NumberResponseItemType {
MOBILE = 'MOBILE',
LANDLINE = 'LANDLINE',
QUICKDIAL = 'QUICKDIAL',
INTERNATIONAL = 'INTERNATIONAL',
}
The mnpState
of type NumberMnpState
only appears in a mobile number entry and looks like this:
interface NumberMnpState {
isReleased: boolean;
releasedUntil: Date;
}
Real Time Call Manipulation (RTCM)
The real time call manipulation module provides the following functions:
interface RTCMModule {
getEstablishedCalls: () => Promise<RTCMCall[]>;
mute: (call: RTCMCall, status: boolean) => Promise<void>;
record: (call: RTCMCall, recordOptions: RecordOptions) => Promise<void>;
announce: (call: RTCMCall, announcement: string) => Promise<void>;
transfer: (call: RTCMCall, transferOptions: TransferOptions) => Promise<void>;
sendDTMF: (call: RTCMCall, sequence: string) => Promise<void>;
hold: (call: RTCMCall, status: boolean) => Promise<void>;
hangUp: (call: RTCMCall) => Promise<void>;
}
The structure of a present call is provided by a RTCMCall
and containts the following attributes:
interface RTCMCall {
callId: string;
muted: boolean;
recording: boolean;
hold: boolean;
participants: Participant[];
}
interface Participant {
participantId: string;
phoneNumber: string;
muted: boolean;
hold: boolean;
owner: boolean;
}
The getEstablishedCalls
method:
It returns all present calls of this account and return an array of RTCMCall
.
The mute
method:
You can pass a RTCMCall
and set your microphone muted or unmuted.
The record
method:
You can start or stop a recording and find this later in the history entry.
The announce
method:
You can play an audiofile that needs to be a mono 16bit PCM WAV file with a sampling rate of 8kHz. Insert the URL of this audiofile as parameter in the announcment.
The transfer
method:
You can attend a call and transfer it to a phonenumber with the following structure:
interface TransferOptions {
attended: boolean;
phoneNumber: string;
}
The sendDTMF
method:
You can send a sequence of valid DTMFs digits to the passed call.
The hold
method:
You can hold and continue a present call.
The hangUp
method:
You can abort or terminate a current call.
Voicemail
The voicemail module provides the following functions:
async function getVoicemails(): Promise<Voicemail[]>;
The getVoicemails
method returns an array of the following type:
export interface Voicemail {
id: string;
alias: string;
belongsToEndpoint: {
extension: string;
type: string;
};
}
Device
The device module provides the following functions:
async function getDevices(webuserId: string): Promise<Device[]>;
The getDevices
method returns an array of the following type:
export type DeviceType = 'REGISTER' | 'MOBILE' | 'EXTERNAL';
export interface ActiveRouting {
id: string;
alias: string;
}
export interface Device {
id: string;
alias: string;
type: DeviceType;
dnd: boolean;
online: boolean;
callerId?: string;
owner?: string;
activePhonelines: ActiveRouting[];
activeGroups: ActiveRouting[];
}
Examples
For some examples on how to use the library, please refer to this repository: sipgateio-node-examples
npx ts-node some_example.ts
Changelog
You can find the changelog here.
Privacy Note
This library sets the following headers for every request made to the sipgate REST-API to obtain statistics about versions currently in use and to differentiate sipgate.io-users from sipgate software:
- X-Sipgate-Client
- X-Sipgate-Version