VoIP Help and Support

SIP, Asterisk, OpenSIPS: Frequently Asked Questions

On this page are links to various articles explaining different aspects of VoIP technology.

Understanding SIP:

Select one of the links below for more information about the SIP protocol:

Can you make free calls using Voice over IP (VoIP)?
How do you choose the best VoIP Service Provider?
SIP Servers explained
SIP Registration explained
What is the "SIP Proxy" setting on an IP phone?
What is the "Outbound Proxy Server" setting on an IP phone?
SIP and NAT: Why is it a problem?
SIP devices behind NAT: What solutions are available?
What options are there for High Availability and Failover with SIP?

Configuring Asterisk:

Smartvox has been working with Asterisk for several years. The help we offer here describes how you configure Asterisk by direct editing of the CONF files. It may be of help to people who are using FreePBX, Trixbox, AsteriskNow or other GUI configuration utilities, but there are no direct references to any of these interfaces. You are welcome to take advantage of our expertise and experience by selecting one of the links below which are arranged as a list of Frequently Asked Questions:

How can an Asterisk IP-PBX benefit my business?
How must Asterisk be configured when it is behind NAT?
What is the SIP Subscribe/Notify mechanisms and how do Asterisk hints relate to it?
What are SIP Domains and how do you use them in Asterisk?
How do you configure Asterisk to support IP phones as extensions?
How do you get Shared Line Appearances (SLA) to work on IP phones with Asterisk?
How can I improve the resilience of my Asterisk service or introduce failover?

SER, OpenSER and OpenSIPS:

Information about the popular Open Source SIP Proxy solution:

Installing OpenSIPS v1.6.1 on CentOS 5 64 bit
Installing Mediaproxy 2 on CentOS 5 64 bit