No subject


Thu Jul 12 09:23:04 CDT 2007


<br>
** Login/Logout of queues, Day/Night mode buttons with indication (1.6<br>
has this as well).<br>
** Company internal directory on the phone updated on the PBX</blockquote><div>&nbsp;Some (most ?) IP phones support this<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
** System Speed Dial on the display updated by the PBX</blockquote><div>This one is interesting.<br>I can&#39;t see a way to do it.<br>Ant idea ? <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
** Call Fwd by PBX with LED indication (not phone based callfwd which sucks).</blockquote><div>Some IP phones support this<br>&nbsp;
<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
** On screen Voicemail (on the phone).</blockquote><div>high end ip phones (XML) should support <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
** Line assignment to buttons with LED indication, and hold indication.</blockquote><div><br>For this one, I don&#39;t know. SCA, maybe ?<br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
** Hold ringback (some IP phones support it).<br>
There are many more features but I can&#39;t remember them at the moment.<br>
<br>
Granted in bigger installations there many more factors and usually<br>
more funding which makes the above list almost obsolete for the<br>
features that Asterisk does have.<br>
<br>
Again my advice do not go with Asterisk for this installation go with Panasonic.<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
<br>
<br>
&gt;&gt; What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys<br>
&gt;&gt; SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured<br>
&gt;&gt; Asterisk/Fedora 9 so I can make SIP-&gt;PSTN and PSTN-&gt;SIP calls.<br>
&gt;&gt;<br>
&gt;&gt; Works. Now, I need this help, please:<br>
&gt;&gt;<br>
&gt;&gt; * Dialing from inside (pap2-FXS connected phone) to another number on<br>
&gt;&gt; the same city (goes out by SPA3102 FXO), voice works fine. But when a<br>
&gt;&gt; menu answers, and I dial over, the menu dialed keys works only 20% of<br>
&gt;&gt; all times. Why could this would be? Voltage levels? sound gains? Dialed<br>
&gt;&gt; keys get distorsioned when passing over the 2 Linksys? Linksys or<br>
&gt;&gt; Asterisk swallowing some dialed key? I noticed some echo...<br>
&gt;&gt;<br>
&gt; Probably you are sending dtmf signals inband. Try outband.<br>
&gt; For the echo, try to change the FXO/FXS impedance, and/or playing with<br>
&gt; the rx and tx gains. I assume that do you have echo cancelling enable in<br>
&gt; both SPA.<br>
&gt;&gt; * I need to assign two codes to each user, one for international calls<br>
&gt;&gt; charged to the office, another for international calls charged to the<br>
&gt;&gt; user. If the user enters an incorrect code, the call should not proceed.<br>
&gt;&gt;<br>
&gt; See account codes. You can start here:<br>
&gt; <a href="http://www.voip-info.org/wiki-Asterisk+Billing" target="_blank">http://www.voip-info.org/wiki-Asterisk+Billing</a><br>
&gt;<br>
&gt;&gt; * I need to get a formatted calls report for the administrators to<br>
&gt;&gt; charge the users.<br>
&gt;&gt;<br>
&gt; See same link, or google for billing<br>
&gt;&gt; I just am confused and stucked with all the documentation in Internet,<br>
&gt;&gt; and all this new asterisk jargon. I just need some links (or some<br>
&gt;&gt; directions) to go fast on this topics. Of course, some more help would<br>
&gt;&gt; be appreciated.<br>
&gt;&gt;<br>
&gt; The link to start:<br>
&gt; <a href="http://www.voip-info.org" target="_blank">http://www.voip-info.org</a><br>
&gt;<br>
&gt;&gt; Thanks a lot.<br>
&gt;&gt;<br>
&gt; De nada<br>
&gt;<br>
&gt; Jorge<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
&gt;<br>
&gt; asterisk-users mailing list<br>
&gt; To UNSUBSCRIBE or update options visit:<br>
&gt; &nbsp; <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
&gt;<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
 &nbsp; <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br></div>

------=_Part_44798_33086575.1224161135545--



More information about the asterisk-users mailing list