What is SIP trunk? (or authorization by IP)

The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, as well as in instant messaging over Internet Protocol (IP) networks.

By enabling SIP trunk, you get a possibility to use any amount of connected numbers as a CallerID without any additional settings to your personal account.

When do you need SIP trunk?

For example, you have a big call center, which has several Internet channels: the main channel and two redundant channels. You can enable SIP trunk and confirm your static IP addresses. Now you can make outgoing calls from these IP addresses only, and there will be no need to enter your login and password every time you reset your hardware to factory settings, or when you log out by accident. Your login/password is not needed anymore to make calls. By setting up the SIP trunk once and connecting new numbers (for example, 10 new numbers), you instantly get access to use them as your CallerIDs without any additional settings added to your personal account.

Please, note:

It is essential to have a static IP address in order to use the SIP trunk. The SIP trunk works on the basis of an already existing SIP login. After enabling SIP trunk, your SIP login becomes unavailable for registration (receiving incoming calls). You can receive incoming calls by using the free SIP login or without registration by SIP URI.

When using SIP trunk, authorization will be made via your IP address (without login/password).

Opening SIP trunk

  • On your personal account under the tab "Settings - SIP settings" at the bottom of the page click "Add".
  • Set the SIP trunk name and choose one of existing SIP logins. This will identify the SIP trunk and the SIP will become unavailable for registration (receiving incoming calls).
  • Add your static IP address or several IP addresses.
  • After IP address is confirmed, it is necessary to route a call through sip.zadarma.com on the number 8888.
  • After your IP address is confirmed, you can make outgoing calls by sending an INVITE to us with the necessary number inserted as a CallerID in the header “From:”. This will make it significantly easier for us to set up, especially if you have a large amount of numbers and it makes it easier to understand the work of outgoing routing.

INVITE sip:4444@sip.zadarma.com SIP/2.0

From: <sip:442037691880@sip.zadarma.com>

To: <sip:4444@sip.zadarma.com>

Number in CallerID in header From: should be sent in international format E.164. It can be one of connected or confirmed numbers. In case if in header From: is sent number in incorrect format, CallerID set by default on SIP login in personal account in SIP trunk settings will be used.

For example, in Asterisk with the help of two lines only in extensions.conf you can set sending of necessary CallerID depending on call destination:

exten => _7495XXXXXXX,1,Set(CALLERID(num)=74951234567)

same => n,Dial(SIP/zadarma/${EXTEN})

*Moscow CallerID 74951234567 will be used for calls to Moscow landline numbers starting with code 7495