Panoramisk / The VoIP druid 

You are now an IP telephony user and have installed an Asterisk. You are using an IP Centrex link to a SIP or IAX operator. Go to the next level with peer-to-peer telephony with ENUM.
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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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It could sometimes be useful to be able to send a vocal message to a group of phone with the intercom feature. Most IP phones on the market support this feature which allows a call to be automatically answered and its content pushed to the speaker.
What we propose in this article is a solution to distribute a live message to a specific phone group. Such solutions are available on the net but since we haven’t found a complete functional one, we have built our own one on top of an Asterisk 1.4, the conference application and an automatic dialler written as an AGI.
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As soon as we install in an organisation an Asterisk IP telephony service, it can impact directly on the service and we should study how it can be resilient. This article proposes a very simple but redundant telephony service with two Asterisk servers installed in a load-balancing mode.
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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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