heeby / sipgate.io

sipgate.io Documentation

sipgate.io

This README documents the sipgate.io functionality. There's a demo page, code examples, and a newsletter.

Requirements

Usage with simquadrat

• Order a simquadrat SIM
• Book the sipgate.io feature
• Enter an URL for incoming/outgoing calls in the dashboard

Usage with sipgate basic

• Create a free sipgate basic account
• Book the sipgate.io feature
• Enter an URL for incoming/outgoing calls in the dashboard

Usage with sipgate team

• Book sipgate.io in your team account (incurs monthly cost) or request access to our developer program (free!)
• After receiving the confirmation mail enter an URL for incoming/outgoing calls in sipgate team settings

The POST request

When enabled, sipgate.io sends POST requests with an application/x-www-form-urlencoded payload for each of call involving your sipgate account. Depending on the type of request it contains the following parameters:

New call

Parameter	Description
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")
event	"newCall"
callId	A unique alphanumeric identifier to match events to specific calls
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.

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

curl -X POST --data "from=492111234567&to=4915791234567&direction=in&event=newCall&callId=123456&user[]=Alice&user[]=Bob" http://localhost:3000

=======

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

Answer

If you set the "onAnswer" attribute sipgate.io will push an answer-event, when a call is answered by the other party.

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

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" http://localhost:3000

Call hangup

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

Parameter	Description
event	"hangup"
cause	The cause for the hangup event (see table below)
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=hangup&cause=normalClearing&callId=123456" http://localhost:3000

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

DTMF

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

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

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.

Dial

Redirect the call and alter your caller id (call charges apply).

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>

Play

Play a given 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>:

• Play 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>

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>

Hangup

Hang up calls

Example: Hang up call

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

onAnswer

Example: Request notification for call being answered

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

onHangup

Example: Request notification for call hangup

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

More to come

Stay tuned...

Code Examples

To get you started we maintain server examples for:

• Node.js
• PHP

There are also examples in:

• C++
• Clojure
• Common Lisp
• Go
• Java
• Perl
• Prolog
• Python
• Ruby
• Scala

Troubleshooting

sipgate.io Log

You can enable logging for debugging purposes from your dashboard. You will find each request and the corresponding response in the logging table.

How do I inspect network traffic?

You can use ngrep to inspect the incoming requests on your side:

sudo ngrep -dany -Wbyline port 3000

A word about security

HTTP vs. HTTPS

We strongly encourage you to use a HTTPS server. Although we support plain HTTP connections we do not recommend pushing sensitive call details over unencrypted connections. By default sipgate.io does not accept self-signed certificates. If you use simquadrat or sipgate basic, you can allow self-signed certs in your dashboard. In sipgate team you cannot make an exception.

Furthermore, you can add the public key of your certificate to protect the connection against man-in-the-middle attacks. Your certificate is validated by our server.

Authentication

sipgate.io supports HTTP Basic Authentication. You can include your username and password within the URL (e.g. https://username:[email protected]:8080).

Emergency calls

sipgate.io does not process emergency calls. Emergency calls are immediately put through to emergency services.

Help us make it better

Please tell us how we can improve sipgate.io. If you have a specific feature request, found a bug or would like to add an example, please use GitHub Issues or fork these docs and send a pull request with your improvements.