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FAQ

What is a SIP trunk? (authorization via 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.

If you create a SIP trunk, you can use any amount of connected phone numbers for the CallerID without any additional setup in your personal profile.

When do you need to use the SIP trunk?

If you have a large call center with multiple Internet channels (one main channel and several back up channels) you can activate the SIP trunk and confirm your IP addresses. In this case, you can only make outgoing calls from the confirmed IP addresses. You will no longer need to enter your login details when you reset your hardware, for example, if the hardware was accidentally reset to its default settings, or if you have logged out by accident. By creating the SIP trunk and connecting new phone numbers, you can use these numbers as your CallerID without the need for any additional setup.

Please, note:

To use the SIP trunk it is essential to have a static IP. After activating the SIP trunk, your SIP login becomes unavailable for registration (receiving incoming calls). Incoming calls can be received using a free SIP login, or without registration using SIP URI.

When using the SIP trunk, authorization is be made via your IP address (without using the login details).

Creating the SIP trunk

  • Go to «Settings», «SIP settings» in your personal profile and click «Add a SIP Trunk».
  • Create a name for the SIP trunk and select one of the existing SIP logins. The selected SIP login will be the identifying SIP for your SIP trunk and will become unavailable for the registration (receiving incoming calls).
  • Add your static IP address or several IP addresses.
  • To confirm the IP addess, it is necessary to route the call to number 8888 on sip.zadarma.com server.
  • After your IP address is confirmed, you can make outgoing calls by sending us an INVITE with the appropriate CallerID number in the «From:» section. This will simplify the work at our end and will help us 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>

The CallerID number in the «From:» section must be in an international format E.164. It can be a phone number purchased from Zadarma or any other confirmed phone number. If the phone number in the «From:» section is entered in the wrong format, we will use the CallerID number set by default in your personal profile’s SIP trunk settings.

For example, if you are using Asterisk, you can set up different CallerIDs depending on your call destination using two lines in extensions.conf:

exten => _4420XXXXXXX,1,Set(CALLERID(num)=442012345678)

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


*London CallerID 442012345678 will be used for calls to London landline numbers starting with code 420