<div dir="ltr"><br><br><div class="gmail_quote">2008/10/16 C F <span dir="ltr"><<a href="mailto:shmaltz@gmail.com">shmaltz@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
* Live call screening - Yes there is a hack that can do it, but it's a<br>
hell of a hack.<br>
* Phones that can do most of the usefull features supported by the PBX<br>
for a reasonable price with LED buttons, including the following<br>
features:<br>
** Call recording with LED indication, while at it, the recordings<br>
integrate seamlessly with your voicemail, which means you don't need<br>
to browse the file system on the PBX to listen to it.</blockquote><div> </div><div>What would be missing to integrate this feature ?<br>With features.conf, it should be possible to map key combinations to an Asterisk application (maybe an AGI script ?)<br>
>From there, it should be possible to drive SIP hardphones BLF status, don't you think ?<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>
** 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> 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'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>
<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'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'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>
>> What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys<br>
>> SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured<br>
>> Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls.<br>
>><br>
>> Works. Now, I need this help, please:<br>
>><br>
>> * Dialing from inside (pap2-FXS connected phone) to another number on<br>
>> the same city (goes out by SPA3102 FXO), voice works fine. But when a<br>
>> menu answers, and I dial over, the menu dialed keys works only 20% of<br>
>> all times. Why could this would be? Voltage levels? sound gains? Dialed<br>
>> keys get distorsioned when passing over the 2 Linksys? Linksys or<br>
>> Asterisk swallowing some dialed key? I noticed some echo...<br>
>><br>
> Probably you are sending dtmf signals inband. Try outband.<br>
> For the echo, try to change the FXO/FXS impedance, and/or playing with<br>
> the rx and tx gains. I assume that do you have echo cancelling enable in<br>
> both SPA.<br>
>> * I need to assign two codes to each user, one for international calls<br>
>> charged to the office, another for international calls charged to the<br>
>> user. If the user enters an incorrect code, the call should not proceed.<br>
>><br>
> See account codes. You can start here:<br>
> <a href="http://www.voip-info.org/wiki-Asterisk+Billing" target="_blank">http://www.voip-info.org/wiki-Asterisk+Billing</a><br>
><br>
>> * I need to get a formatted calls report for the administrators to<br>
>> charge the users.<br>
>><br>
> See same link, or google for billing<br>
>> I just am confused and stucked with all the documentation in Internet,<br>
>> and all this new asterisk jargon. I just need some links (or some<br>
>> directions) to go fast on this topics. Of course, some more help would<br>
>> be appreciated.<br>
>><br>
> The link to start:<br>
> <a href="http://www.voip-info.org" target="_blank">http://www.voip-info.org</a><br>
><br>
>> Thanks a lot.<br>
>><br>
> De nada<br>
><br>
> Jorge<br>
><br>
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