OpenSIPS is an open source and free SIP server. It can be used to manage your VoIP network, or you can use it as a software development kit for creating your own VoIP applications.

This is written in C and has been developed in order to provide a high quality, reliable and flexible framework for developing applications that require any kind of VoIP management. OpenSIPS is not only a SIP server but also a software development kit which allows you to create your own VoIP applications.

What is OpenSIPS?

OpenSIPS is a free and open-source SIP server. It allows you to run your own SIP service provider.

OpenSIPS is an open-source, free SIP server that enables you to run your own service provider. The project was originally created in 2004 by Igor Novikov and it was released under the GPLv2 license. OpenSIPS is written in C language.

The latest release of OpenSIPS is version 2.2 and it has been published on January 19th, 2018.

How to Install OpenSIPS on Ubuntu

This tutorial will show you how to install OpenSIPS on Ubuntu. OpenSIPS is a free and open-source SIP server that can be deployed in both embedded and standalone configurations.

The first step is to download the latest version of the installation package from the official website.

In this example, we are going to use the latest version of OpenSIPS for Ubuntu 16.04 LTS (xenial).

  1. Once downloaded, extract it by running: tar xzf open siips-1.5-1.tar.gz
  2. Next, change directory into the extracted folder by running: cd open siips-1.5-1
  3. Now run: sudo. /install

SIP Configuration Options for OpenSIPS

There are a number of options that can be configured when configuring the OpenSIPS server. Each option is set in a different location and some options are set multiple times.

The first configuration option is to choose which type of interface to use, either UDP or TCP. The next configuration option is to configure the IP address of the interface. The third configuration option is to configure what port the SIP server should listen on. The fourth configuration option is whether or not TLS should be enabled for encryption, with an optional password for TLS encryption if desired.

The next few configuration options are for logging purposes, with one being for logging general information about OpenSIPS and another being for logging errors from OpenSIPS. There are also three different types of logs that can be configured: syslog, file, and memory-based log files.

Creating Users and Groups for Voicemail and Presence in SipXecs

Voicemail and Presence are two of the most important features of a VoIP service. Voicemail is a feature that allows users to leave messages for one another, while Presence is a feature that keeps track of user availability. Read more about VoIP service..

Configuring Groups:

SipXecs allows you to create groups for different departments or teams, so that you can assign different privileges to each group. For example, if you want to create an "Operations" group, then the members of this group will have access to voicemail and presence features. You can also configure which group has access to which features. For instance, if you want your "Operations" group to only have access to voicemail and not presence then all they would need is voicemail privileges configured under their account settings.

Creating Users:

To create a SipXecs user account, navigate over to the "User Management" tab on your left hand side and click on "Add User". You

Configuring SS7 Routing to Use a PSTN Gateway in OpenSIPS

The PSTN Gateway is a device that has been developed to provide SS7 routing between a PSTN network and a VoIP network.

Configuring OpenSIPS to use the PSTN Gateway is not difficult. The first step is to configure the gateway with your IP address and credentials. The second step is to configure the SIP trunk in OpenSIPS by supplying the gateway as an outbound proxy server.

The third step would be to configure the VoIP trunk in OpenSIPS by supplying the gateway as an outbound proxy server. This would be done for both directions of communication, so that voice traffic from your VoIP network can reach your PSTN network, and vice versa.


OpenSIPS is an open source and free SIP server that can be used for routing and proxying voice, video, and data. It is also a very flexible system that can be easily customized with scripts and configured for specific needs.



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