Author: John

  • E1

    E1 E1 is an ITU-T standard that defines the basic protocol and structure for transmission of TDM data over a physical circuit (most often the physical connections use twisted pairs and conform to G.703). It is widely used in Europe. An E1 primary rate trunk has a total of 32 time slots allocated within each frame of data. In TDM telephony applications, time slot 16 is used as a data channel (the D channel) and carries call setup, clear down and…

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  • TDM

    TDM Stands for: Time Division Multiplexed This is one method by which multiple streams of data are combined together into a single stream of data for transmission down a cable or optical fibre. The combined streams may then be separated out again into individual streams at the far end of the cable as required. When digital data are merged using time division multiplexing, the resulting merged data stream is structured as a series of small blocks, called…

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  • How secure is your Asterisk PBX? – part 3

    Getting more advanced In part 2, we looked at several ways in which an Asterisk system administrator can help to make their system more secure, with special emphasis on avoidance of toll fraud. In this, the third and final article in the series, I will pick up on a topic that was left unfinished at the end of part 2 – sip domains. I also want to look at a couple of other topics that were barely touched…

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  • How secure is your Asterisk PBX? – part 2

    Protecting your Asterisk server In part 1, we examined the techniques that are used to probe for vulnerabilities in a SIP server and reviewed the types of exploitation a would-be hacker hopes to use. In this second part, I look at the ways you can protect your Asterisk or other SIP server and guard against weaknesses that could potentially cost your organisation a lot of money. This article was written before the pjsip channel driver was available,…

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  • How secure is your Asterisk PBX? – part 1

    A growing problem Like a slice of Victoria sponge cake on a summers day attracts wasps, so new technologies seem to attract the attention of cyber-criminals. The more widely used the technology, the greater the interest. It was inevitable, and widely predicted, that VoIP would become a favorite target for hackers as its popularity and uptake increased – it has the accessibility of an email server combined with the potential for fraud of an online bank account. Irresistible! And so…

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  • Installing Mediaproxy 2 on CentOS 5 64bit

    Mediaproxy 2 is a Python application from AG-Projects which is available as a free download as well as being available as a commercial product from AG-Projects. It is used in combination with the Mediaproxy module of OpenSIPS. Unlike Mediaproxy 1, the new version has a lot of dependencies and it can be quite tricky to install. The INSTALL instructions that come with the package are very helpful, but unfortunately they are aimed primarily at installers who…

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  • IP Phone Configuration: User Account Settings

    Manufacturers different naming conventions While SIP is an industry standard protocol, the names assigned to the configuration fields on an IP phone are far from standardised. The tables below show the text labels that you can expect to see on the web interface configuration forms next to the boxes where you must enter your settings. We show examples of the labels used for equivalent fields by four manufacturers: Snom, Aastra, Grandstream and Linksys. User accounts There…

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  • Installing OpenSIPS v1.6.1 on CentOS 5 64bit

    Smartvox Limited offered consultancy for installation and configuration of OpenSIPS, Asterisk and PSTN gateways. If you want to do it all yourself, here is a set of instructions that should help. Installing OpenSIPS v1.6.1 on CentOS 5 64 bit is very straightforward. All the dependencies can be installed using YUM. The packages required will depend on the modules that you intend to use from OpenSIPS. This page describes what must be installed to use OpenSIPS with…

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  • 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 device. The remote device that is connecting to Asterisk is behind NAT Suppose that your Asterisk server is connected directly to…

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  • 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 facilities designed to help them analyse their own networking environment and to help overcome the problems of NAT traversal. Probably the…

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