Panoramisk / The VoIP druid 

Is the Chicago meeting room available? How many time a year do you ear this kind of question in your organization? On top, it is frequent you need a free meeting room for something not planned in advance and your are checking physically all the rooms in the building! In this article we present a solution, based on Asterisk, able to manage the free status of a meeting room with your PBX and line supervision feature.
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Configuring an Asterisk server can be cumbersome for some administrators not used to unix and text file format. There are some solutions to ease this configuration process, mainly on integrated solution where a web interface is proposed. Apstel is currently proposing another solution which is interesting: « Visual Dialplan » since it is mainly based on a graphical view of the call plan.
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In order to enable voice call exchange between two PBX the straightforward protocol to use is SIP. It can allows device interconnection, regardless of their respective role, phone, proxy or gateway. This article explains how simply interconnecting two Asterisk PBX and could be used as a template in order to configure a SIP trunk between an Asterisk PBX and any other one available on the market.
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When interconnecting two PBX, it can be useful to transport special information in addition to standard call signalling messages. When using such an open protocol as SIP, which supports parameters inclusion in INVITE messages, we can imagine new features for our open telephony. Here are some tips for doing this with an Asterisk solution.
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We do release today our first version of the Asterisk::Configure Perl module, useful for configuring dial plan for Asterisk without having to renumber lines. As a first version, we are looking for feedback in order to improve the library and correct bugs if any.