<div>Your are right, I dont have to invent the wheel again, and I'm getting cleverer by looking at other peoples code.</div>
<div> </div>
<div>But this does not solve my problems, I have worked in the PABX business as a software developer for about 8 years, and coming to * is not all that easy. </div>
<div> </div>
<div>For instance, * does not give you very good information of the state of extensions (like we are used to in the "old-fashioned" PABX business), or maybe I'm not good at finding the information.</div>
<div> </div>
<div>I'm trying to port an existing Windows application to *, its a dialer, used to dial and se information about received calls.</div>
<div> </div>
<div>I know how to dial new calls, by using ORIGINATE on the AMI.</div>
<div>I can receive some status information via the AMI, but consider this example:</div>
<div> </div>
<div>I receive a call, which I accept. I get an event from the AMI, that the call is now in the UP state.</div>
<div>I receive another call, I get en event from the AMI, that the new call is in the RINGING state.</div>
<div> </div>
<div>So far, so good.</div>
<div> </div>
<div>I now answer the other call (for instance by the line button on my phone).</div>
<div>Both calls are now in the UP state, who am I talking to?</div>
<div> </div>
<div>This, and many other questions, are currently making me even more thin haired than normal :-)</div>
<div> </div>
<div> </div>
<div>Michael <br><br> </div>
<div><span class="gmail_quote">2006/8/25, C F <<a href="mailto:shmaltz@gmail.com">shmaltz@gmail.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">So how about inventing a car? The auto industry is much more profitable.<br><br>The point; there is no point in reinventing the wheel, why are you
<br>writing this from scratch?<br><br>On 8/24/06, Mir <<a href="mailto:michael.sysdba@gmail.com">michael.sysdba@gmail.com</a>> wrote:<br>><br>> What do you mean?<br>><br>> I'm not looking for someone elses work, I'm developing an application from
<br>> scratch.<br>><br>> Michael<br>><br>><br>> 2006/8/24, Andrew Kirch <<a href="mailto:AKirch@allthingsit.com">AKirch@allthingsit.com</a>>:<br>> ><br>><br>><br>><br>><br>><br>
> Umm… Flash operator panel?<br>><br>><br>><br>> Andrew<br>><br>><br>><br>> ________________________________<br>><br>><br>> From: <a href="mailto:asterisk-dev-bounces@lists.digium.com">
asterisk-dev-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a>] On Behalf Of<br>> Mir<br>> Sent: Thursday, August 24, 2006 2:18 PM
<br>> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>; <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br>> Subject: [asterisk-dev] Phone status
<br>><br>><br>><br>><br>><br>> Hi<br>><br>><br>><br>><br>><br>> I'm working on a project, where I need the status of every telephone on the<br>> system. (Idle,ringing,busy)<br>><br>
><br>> If a phone is busy, I also need to know the callerid of the other end.<br>><br>><br>><br>><br>><br>> I have made a deamon, which query Asterisk every second for active calls,<br>> this works by issuing a "Status" to the manager-interface, and processing
<br>> the return data and then put the result into a MySQL table.<br>><br>><br>><br>><br>><br>> The clients will query the MySQL table every second for the state of their<br>> phone, if there are no records with their numbers in it, they are considered
<br>> idle.<br>><br>><br>><br>><br>><br>> This works fine for calls from one SIP-phone to the other, this is for<br>> instance what it look like when extension 310 is connected to extension 311:<br>
><br>><br>><br>><br>><br>> Event: Status<br>> Privilege: Call<br>> Channel: SIP/310-08697fb8<br>> CallerID: 310<br>> CallerIDName: <unknown><br>> Account:<br>> State: Up<br>> Link: SIP/311-0868fd98
<br>> Uniqueid: 1156442804.74<br>><br>><br>> Event: Status<br>> Privilege: Call<br>> Channel: SIP/311-0868fd98<br>> CallerID: 311<br>> CallerIDName: Snom<br>> Account:<br>> State: Up<br>> Context: macro-vm
<br>> Extension: s<br>> Priority: 5<br>> Seconds: 13<br>> Link: SIP/310-08697fb8<br>> Uniqueid: 1156442804.73<br>><br>> That is pretty easy to decode.<br>><br>> However when an external call is made to a SIP-phone, the result is
<br>> different, this is a call from another Asterisk via an IAX trunk:<br>><br>> Event: Status<br>> Privilege: Call<br>> Channel: SIP/311-08695698<br>> CallerID: 35254390<br>> CallerIDName: <unknown>
<br>> Account:<br>> State: Up<br>> Link: IAX2/MR-1<br>> Uniqueid: 1156442974.76<br>><br>><br>> Event: Status<br>> Privilege: Call<br>> Channel: IAX2/MR-1<br>> CallerID: 35436121<br>> CallerIDName: <unknown>
<br>> Account:<br>> State: Up<br>> Context: macro-vm<br>> Extension: s<br>> Priority: 5<br>> Seconds: 9<br>> Link: SIP/311-08695698<br>> Uniqueid: 1156442974.75<br>><br>> The actual callerid of the caller is 3536121, 35254390 is the called number.
<br>><br>> How do I get the information, that 35436121 is connected to 311?<br>><br>> Am I doing it in a stupid way, I'm aware that the Manager can give me<br>> realtime events, but I'm under the impression, that it is not very stable in
<br>> a high traffic environment?<br>><br>> Any help or good ideas would be appriceated.<br>><br>> Michael<br>><br>><br>><br>><br>><br>><br>><br>><br>><br>> _______________________________________________
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