<div>Ciao,</div>
<div>may be is enough the Free PBX admi web installed with standard Asterisk now distribution.</div>
<div>Real time status is for sure available.<br><br></div>
<div class="gmail_quote">2009/11/11 Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>></span><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi Christina!<br><br> From documentation it seems it only supports queues and agents. I do<br>not have a single queue nor agents. Does it also support real-time<br>
status for normal SIP-SIP calls?<br><br>regards<br>klaus<br><br>Christina Casey schrieb:<br>
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<div class="h5">> Hi Klaus,<br>><br>> Yes all the below is possible/easy with the OrderlyStats call centre<br>> management and reporting tool.<br>><br>> It's a free download - please see <a href="http://www.orderlyq.com/orderlystats.html" target="_blank">http://www.orderlyq.com/orderlystats.html</a><br>
><br>> Kind regards,<br>><br>> Christina Casey<br>> Accounts Manager<br>> Orderly Software Ltd.<br>><br>><br>>><br>>> Subject:<br>>> [asterisk-users] looking for an Asterisk supervision (status viewer) tool<br>
>> From:<br>>> Klaus Darilion <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>><br>>> Date:<br>>> Tue, 10 Nov 2009 14:04:16 +0100<br>>> To:<br>>> Asterisk Users Mailing List - Non-Commercial Discussion<br>
>> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>>><br>>> To:<br>>> Asterisk Users Mailing List - Non-Commercial Discussion<br>>> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
>><br>>><br>>> Hi!<br>>><br>>> I am looking for a tool (application or webinterface) which shows me<br>>> the current status of an Asterisk server, e.g.:<br>>><br>>> - Status of the SIP peers (registered/offline)<br>
>> - current incoming and outgoing calls<br>>> - start-time, numbers, some history<br>>> - history (calls stopped in the last 15 minutes, who hang up?)<br>>> - should be possible to link those calls to the relevant SIP peers<br>
>> - "kill" calls<br>>><br>>> Before coding it myself, is there something you can recommend to me?<br>>><br>>> The thing should be complete auto configured, e.g. no configuration<br>
>> file which peers/channels to be displayed, just fetch all the<br>>> configuration from Asterisk and display it.<br>>><br>>> thanks<br>>> klaus<br>>><br>>><br>><br>><br></div>
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