API References

Parameter	Description
event	„newCall“
from	The calling number (e.g. „492111234567“ or „anonymous““)
to	The called number (e.g. „492111234567“)
direction	The direction of the call (either „in“ or „out“)
callId	A unique alphanumeric identifier to match events to specific calls.
origCallId	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. It is always „user[]“ (not „user“), even if only one user is involved.
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

Optional Parameter	Description
diversion	If a call was diverted before it reached sipgate.io this contains the originally dialed number.

The XML response

After sending the POST request sipgate.io will accept an XML response to determine what to do. Make sure to set .application/xml in the Content-Type header of your response.

sipgate.io currently supports the following responses for incoming and outgoing calls:

Action	Description
Dial	Send call to voicemail or external number
Play	Play a sound file
Gather	Collects digits that a caller enters with the telephone keypad.
Reject	Reject call or pretend to be busy
Hangup	Hang up the call

Actions

Dial

Redirect the call and alter your caller id (call charges apply). Calls with direction=in can be redirected to up to 5 targets.

Attribute	Possible values	Default value
callerId	Number in E.164 format	Account settings
anonymous	true, false	Account / phone settings

Possible targets for the dial command:

Target	Description
Number	Send call to an external number (has to be in E.164 format)
Voicemail	Send call to voicemail (feature has to be booked)

Example 1: Redirect call

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Dial>
        <Number>4915799912345</Number>
    </Dial>
</Response>

Example 2: Send call to voicemail

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Dial>
        <Voicemail />
    </Dial>
</Response>

Example 3: Suppress phone number

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Dial anonymous="true">
        <Number>4915799912345</Number>
    </Dial>
</Response>

Example 4: Set custom caller id for outgoing call

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Dial callerId="492111234567">
        <!-- Originally dialed number, extracted from POST request -->
        <Number>4915799912345</Number>
    </Dial>
</Response>

Example 5: Redirect incoming call to multiple destinations

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Dial>
        <Number>4915799912345</Number>
        <Number>492111234567</Number>
    </Dial>
</Response>

When the call is answered, the resulting answer-event reports the answering destination in a field called answeringNumber.

Play

Play a given sound file. Afterwards the call is delivered as it would have been without playing the sound file.

Target	Description
Url	Play a sound file from a given URL

Example 1: Play a sound file

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Play>
        <Url>http://example.com/example.wav</Url>
    </Play>
</Response>

Please 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

Gather

Gather collects digits that a caller enters with the telephone keypad. The onData attribute is mandatory and takes an absolute URL as a value.

Attribute	Possible values	Default value
type	dtmf	dtmf
onData	absolute URL	–
maxDigits	integer >= 1	1
timeout (ms)	integer >= 1	2000

Nesting Rules

The following verbs can be nested within <Gather>:

The timeout starts after the sound file has finished playing. After the first digit is received the audio will stop playing.

Example 1: DTMF with sound file and additional parameters

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Gather onData="http://localhost:3000/dtmf" maxDigits="3" timeout="10000">
        <Play>
            <Url> https://example.com/example.wav</Url>
        </Play>
    </Gather>
</Response>

A configured gather action always triggers a push event, regardless if the callee sent any DTMF data or not. The push contains following values:

Key	Value
event	The string ‚dtmf‘
callId	Unique identifier of the running call, e.g. ‚54566B150B0C0D3A5D544C‘
dtmf	The DTMF data sent by the callee, e.g. ‚1234‘, completely ommited if the callee sent no data

Reject

Pretend to be busy or block unwanted calls.

Attribute	Possible values	Default value
reason	rejected, busy	rejected

Example 1: Reject call

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Reject />
</Response>

Example 2: Reject call signaling busy

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Reject reason="busy" />
</Response>

Example: Hang up calls

<?xml version="1.0" encoding="UTF-8"?>
<Response>
    <Hangup />
</Response>

Following events

Additional to actions, the response can specify urls which shall be called by sipgate.io on certain call-events. Specify these urls via xml-attributes in the response-tag.

Url	Description
onAnswer	Receives a POST-request as soon as someone answers the call. The response to that request is discarded.
onHangup	Receives a POST-request as soon as the call ends for whatever reason. The response to that request is discarded.

onAnswer

If you set the onAnswer attribute sipgate.io will push an answer-event, when a call is answered by the other party. You will also receive an answer-event if your voicemail is active.

Example: Request notification for call being answered

<?xml version="1.0" encoding="UTF-8"?>
<Response onAnswer="http://localhost:3000/answer" />

Parameter	Description
event	„answer“
callId	Same as in newCall-event for a specific call
user	Name of the user who answered this call. Only incoming calls can have this parameter
userId	The ID of sipgate user(s) involved (e.g. w0).
fullUserId	The full ID of sipgate user(s) involved (e.g. 1234567w0).
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“)
answeringNumber	The number of the answering destination. Useful when redirecting to multiple destinations

You can simulate this POST request and test your server with a cURL command:

curl \
  -X POST \
  --data "event=answer&callId=123456&user=John+Doe&userId=w0&fullUserId=1234567w0&from=492111234567&to=4915791234567&direction=in&answeringNumber=21199999999" \
  http://localhost:3000

Optional Parameter	Description
diversion	If a call was diverted before it reached sipgate.io this contains the originally dialed number.

onHangup

If you set the onHangup attribute sipgate.io will push a hangup-event when the call ends.

Example: Request notification for call hangup

<?xml version="1.0" encoding="UTF-8"?>
<Response onHangup="http://localhost:3000/hangup" />

Parameter	Description
event	„hangup“
cause	The cause for the hangup event (see table below)
callId	Same as in newCall-event for a specific 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“)
answeringNumber	The number of the answering destination. Useful when redirecting to multiple destinations

You can simulate this POST request and test your server with a cURL command:

curl \
  -X POST \
  --data "event=hangup&cause=normalClearing&callId=123456&from=492111234567&to=4915791234567&direction=in&answeringNumber=4921199999999" \
  http://localhost:3000

Optional Parameter	Description
diversion	If a call was diverted before it reached sipgate.io this contains the originally dialed number.

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

onData

If you „gather“ users‘ dtmf reactions, this result is pushed as an event to the url specified in the onData attribute with the following parameters:

Parameter	Description
event	„dtmf“
dtmf	Digit(s) the user has entered. If no input is received, the value of dtmf will be empty.
callId	Same as in newCall-event for a specific call

You can simulate this POST request and test your server with a cURL command:

curl \
  -X POST \
  --data "event=dtmf&dtmf=1&callId=123456" \
  http://localhost:3000

Advanced scenarios

In addition to answering push requests synchronously you can interact with calls in real time through our RTCM-API. Use the callId parameter that is included in every push request.

API References

The POST request

Follow up events

The XML response

Actions

Dial

Play

Gather

Nesting Rules

Reject

Following events

onAnswer

onHangup

onData

Advanced scenarios