<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> > _______________________________________________<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> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> ><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> <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--