Asterisk behind NAT

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…

Using SIP Devices behind NAT

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…

Caller ID in SIP and Asterisk – Part 1

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…

Taking the plunge with SIP Trunks – Part 3

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…

Taking the plunge with SIP Trunks – Part 2

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…

RTP pass-through mode in Asterisk

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…