Category: Legacy Asterisk
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Setting up shared voicemail on Asterisk – part 2
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Part 1 laid the foundations for creating and accessing a shared voicemail box. In this, part 2, I explain how the lamp on the BLF key is switched on and off to show there are messages waiting in the shared box. Note that this is separate from any existing MWI lamp used for personal voicemail. It uses a custom device state within Asterisk switched by an external application. The external application in this case is a bash…
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Setting up shared voicemail on Asterisk – part 1
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It’s a requirement that people often seem to ask for – a single voicemail box, taking messages for a department, that can be easily monitored and accessed by several different users. A typical application would be to record out-of-hours messages which are then checked in the morning by any of a number of users, perhaps just depending who arrives first at the office. While it is fairly easy to configure Asterisk or Trixbox to record the messages, it is less…
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Using Custom Device States to control BLF lamps
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Do you want to know how to use a custom device state to control the lamp on a programmable key of an IP phone? In this article I explain how to set up the hints and make any number of IP phones subscribe to a custom device state and how to switch the custom status from within the Asterisk dial plan. In later articles I plan to show how this can be put to a practical use. I will explain how to…
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How secure is your Asterisk PBX? – part 3
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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
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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
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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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Asterisk behind NAT
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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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Taking the plunge with SIP Trunks – Part 3
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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 call settings shown in part 2. Caveat – this only refers to chan_sip and not pjsip. You will need a peer…
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Taking the plunge with SIP Trunks – Part 2
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Recap In part 1, I explained how a SIP Trunk is really just a virtual connection between your IP-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 DID number. Basic requirements to enable your SIP trunk A SIP Trunk may be used for both inbound and outbound call traffic. To make outbound calls, your Asterisk, FreePBX…
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Taking the plunge with SIP Trunks – Part 1
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Many VoIP service providers offer SIP Trunks 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 or FreePBX? I will attempt to answer these questions in this 3 part blog, starting with the basics. What is a SIP Trunk? A SIP trunk uses standard SIP signalling, but the endpoints of the trunk are fixed – typically one end…
