Smart solutions to VoIP questions
Scenarios in which NAT may adversely affect Asterisk SIP connections The Asterisk Server is behind NAT The Asterisk server could be on the LAN (or in a DMZ) with a NAT firewall between it and the Internet. When it communicates with external peers or devices, the network connections have to pass through the local NAT … Read more
SIP Devices behind NAT: What solutions are available? When an IP phone is installed behind NAT, problems can be created by the NAT device itself, by the phone’s inability to correctly understand its own networking environment or from a combination of the two. Because it is such a common problem, most IP Phones have built-in … Read more
Why is it a problem using SIP Clients behind NAT? What is NAT? To understand why SIP Clients behind NAT are a problem, you need to first have some understanding of what NAT is and what it does. NAT stands for Network Address Translation. Unless you are using One-to-one NAT, then a NAT device may … Read more
Caller ID, ANI, PAI, RPI, Privacy – what Asterisk does I didn’t plan on writing a “Part 2” to this article, but since nobody has posted any answers to my questions in Part 1, I had no option other than to test it for myself. In some ways, the answers are quite simple, but there … Read more
Caller ID, ANI, PAI, RPI, Privacy – can Asterisk cope? Equipment receiving calls, whether a humble handset or a sophisticated Call centre ACD system, likes to know the identity of the caller. It may simply display the caller’s number on an LCD display, look it up in a directory so the caller’s name can be … Read more
Outbound calls; Matching a DDI/DID; Diagnosing problems; Internet bandwidth. Configuration for Outbound calling Part 2 looked only at the configuration for receiving inbound calls, but the SIP Trunk configuration form in Trixbox/FreePBX has to also include the settings for making outbound calls. It may not even work at all if you only use the inbound … Read more
Recap In part 1, I explained how a SIP Trunk is really just a virtual connection between your Asterisk PBX and the VoIP service provider. Standard SIP signalling is used on the trunk, but more than one simultaneous call is allowed and you may have more than one DDI number. Basic requirements to enable your … Read more
SIP Trunks are often talked about on the Asterisk wiki and forum pages; Many VoIP service providers offer them as a standard product; But just what is a SIP trunk and what can you use it for and how do you make it work with Asterisk? I will attempt to answer these questions in what … Read more
What is RTP pass-through mode in Asterisk? The speech for VoIP calls uses RTP (Real Time Protocol) to get from one end to the other and it is compressed using one of the many speech compression codecs available. Commonly used codecs include G.711 and G.729. When you call someone through an Asterisk PBX, there will be … Read more
In a Nutshell: The benefits of upgrading to an Asterisk based IP-PBX