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Caller ID in SIP and Asterisk – Part 2

May 28, 2010

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 […]

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Caller ID in SIP and Asterisk – Part 1

April 20, 2010

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 […]

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Taking the plunge with SIP Trunks – Part 3

December 19, 2009

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 […]

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Taking the plunge with SIP Trunks – Part 2

November 13, 2009

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 […]

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Taking the plunge with SIP Trunks – Part 1

November 1, 2009

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 […]

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RTP pass-through mode in Asterisk

October 23, 2009

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 […]

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How can an Asterisk IP-PBX benefit my business?

August 23, 2009

In a Nutshell: The benefits of upgrading to an Asterisk based IP-PBX

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High Availability and Failover options for SIP and Asterisk

May 29, 2009

Overview What’s the disaster we are trying to avoid? The assumed scenario is this: Some kind of centralised VoIP service is being offered to a number of users; the service operates on servers located at a data centre or office and the users each have a SIP client device, such as an IP phone, that […]

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SIP Subscribe/Notify and Asterisk Hints Explained

March 6, 2009

The SIP SUBSCRIBE/NOTIFY mechanism – what it is and how it works The SIP protocol includes a standardised mechanism to allow any SIP client (an IP phone being an example of a SIP client) to monitor the state of another device. Details are provided in the SIP protocol document RFC 3265. Basically, it works like […]

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Configuring IP phones as Asterisk PBX extensions

February 22, 2009

A first look at the SIP.CONF file It is assumed here that your IP phones all use the SIP protocol to register, make and receive calls. Before an IP phone can connect to Asterisk and operate as an extension, it is necessary to configure user account details on the Asterisk server. The configuration file of […]

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