2008/10/16 C F <[EMAIL PROTECTED]> > > * Live call screening - Yes there is a hack that can do it, but it's a > hell of a hack. > * Phones that can do most of the usefull features supported by the PBX > for a reasonable price with LED buttons, including the following > features: > ** Call recording with LED indication, while at it, the recordings > integrate seamlessly with your voicemail, which means you don't need > to browse the file system on the PBX to listen to it.
What would be missing to integrate this feature ? With features.conf, it should be possible to map key combinations to an Asterisk application (maybe an AGI script ?) >From there, it should be possible to drive SIP hardphones BLF status, don't you think ? > > ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 > has this as well). > ** Company internal directory on the phone updated on the PBX Some (most ?) IP phones support this > > ** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? > > ** Call Fwd by PBX with LED indication (not phone based callfwd which > sucks). Some IP phones support this > > ** On screen Voicemail (on the phone). high end ip phones (XML) should support > > ** Line assignment to buttons with LED indication, and hold indication. For this one, I don't know. SCA, maybe ? > > ** Hold ringback (some IP phones support it). > There are many more features but I can't remember them at the moment. > > Granted in bigger installations there many more factors and usually > more funding which makes the above list almost obsolete for the > features that Asterisk does have. > > Again my advice do not go with Asterisk for this installation go with > Panasonic. > > > > > >> What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys > >> SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured > >> Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls. > >> > >> Works. Now, I need this help, please: > >> > >> * Dialing from inside (pap2-FXS connected phone) to another number on > >> the same city (goes out by SPA3102 FXO), voice works fine. But when a > >> menu answers, and I dial over, the menu dialed keys works only 20% of > >> all times. Why could this would be? Voltage levels? sound gains? Dialed > >> keys get distorsioned when passing over the 2 Linksys? Linksys or > >> Asterisk swallowing some dialed key? I noticed some echo... > >> > > Probably you are sending dtmf signals inband. Try outband. > > For the echo, try to change the FXO/FXS impedance, and/or playing with > > the rx and tx gains. I assume that do you have echo cancelling enable in > > both SPA. > >> * I need to assign two codes to each user, one for international calls > >> charged to the office, another for international calls charged to the > >> user. If the user enters an incorrect code, the call should not proceed. > >> > > See account codes. You can start here: > > http://www.voip-info.org/wiki-Asterisk+Billing > > > >> * I need to get a formatted calls report for the administrators to > >> charge the users. > >> > > See same link, or google for billing > >> I just am confused and stucked with all the documentation in Internet, > >> and all this new asterisk jargon. I just need some links (or some > >> directions) to go fast on this topics. Of course, some more help would > >> be appreciated. > >> > > The link to start: > > http://www.voip-info.org > > > >> Thanks a lot. > >> > > De nada > > > > Jorge > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
