Am Mittwoch, den 15.10.2008, 21:03 -0400 schrieb C F: > On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza <[EMAIL PROTECTED]> wrote: > > Rodolfo Alcazar Portillo wrote: > >> Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in > >> a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can > >> emulate some Panasonic functions on Asterisk fast, to convince the > >> executives. > > Asterisk is more featured than Panasonic, but you must to know Asterisk > > to convince your executives.... ;-) > Not really so. Depending on lots of factors, usually for a small > office of only 5-10 users Panasonic is more feature rich. Since the > main feature they are looking for in a PBX is to be able to yell > across the hallway; "hey boss call on 5 it's your wife" which is not > really possible with Asterisk (yeah I know call parking, but how many > phones support it flawlessly with flashing LEDs?). > Other features that are quite popular in small offices and not > supported by Asterisk: > * 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. > ** 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 > ** System Speed Dial on the display updated by the PBX > ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). > ** On screen Voicemail (on the phone). > ** Line assignment to buttons with LED indication, and hold indication. > ** Hold ringback (some IP phones support it). > There are many more features but I can't remember them at the moment.
Ok, those are to consider, thanks for being specific. Negatives, for me: Forwarding is an important issue for us. I'll read more, search for equivalent equipment before taking the decision. The same with line assignment to buttons. Ok, for me: Screening: do not need it. LEDs: Due to internal policies, we usually buy the essential, so we have just 1 phone with leds, the operator's. I'll buy the best phone for the operator. The rest must be handled manually by her. No queues. No problem with directories. Voicemail find I better on *. The rest, we will suffer, not important for us. > 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. Maybe this is the time for us to switch to *. In some point we must start this new tech. Anyway, thanks for the advice. Rodolfo. > >> 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 -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] _______________________________________________ -- 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
