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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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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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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From version 1.4.4 and revision 69775 the call-limit parameter on a SIP trunk is taken into account.

Maybe have you tried to limit call number on a SIP peer without success, effectivelly the parameter call-limit was not checked, more precisely the number of calls was not updated. This was working for a telephone but not for a peer.

In the sip.conf file, you can use such configuration:

[trunk-test-test2]
type=peer
host=asterisk-test2
qualify=50
context=from-asterisk-test2
username=trunk-test-test2
md5secret=cee0374044a6c33313e1723c79203ad8
call-limit=5

In order to check the current amount of calls, you can issue the command sip show inuse, the result for the above peer with one engaged call is:

* Peer name               In use          Limit
trunk-test-test2          1/0             5

Hope this helps.