hi: i have discussed the transfer function with grandstream engineers. their operation procedure is complicated(eg: no attended+blind transfer). i tell them the simple way, but got no response since then.
Regards, tbskyd 2008/1/7, Andrew Joakimsen <[EMAIL PROTECTED]>: > Another thing is I've found the grandstream phone way of doing things > like transfer, etc much easier to understand for laypeople than the > more expensive phones. There is no clutter of keys or menus. > > On Dec 23, 2007 11:50 AM, d tbsky <[EMAIL PROTECTED]> wrote: > > hi: > > thanks a lot for so many great information. i tried to read the > > specs and manuals for all the phones mentioned. > > we use alcatel pbx in most offices. i surveyed some users to > > understand what functions they use most. and i found few people know > > how to use 3way-conf or forward.i think if the > > function needs two or more keys to operate, then people tend to ignore > > it unless he use that function for daily business. > > i conclude the functions we need are all basic functions. but due > > to the difference of ip pbx/phones and classic pbx/phones, some of > > these functions seem not so "basic" in the ip world: > > > > 1. dial out name display. when you dial a number, the phone lcd will > > show the corresponding name, so you can realize if it is the correct > > number immediately. this needs a corporate directory support, or put > > the whole corporate phonebooks to every ip phone. most ip phone has > > less than 500 local phonebook entries. this is not enough for us. > > grandstream: has xml phonebook support and can combine with local > > phonebooks. > > linksys: has coporate directory but seems only work with linksys > > pbx, not asterisk. > > aastra: has xml phonebook > > snom: has ldap and xml phonebook. xml seems for browsing,don't > > know if work here. > > other china brand phone: none. > > > > 2. transfer. transfer is simple and straightforward in classic pbx. > > you just press "transfer" then dial number and you are on the way of > > attended transfer. you press "transfer" again to cancel transfer. you > > hangup to complete the attended transfer. if you hangup before the > > completion of attended transfer, the transfer becomes blind transfer > > automatically. eventually user didn't notice the "blind" or > > "attended" concept in classic pbx. > > snom: has "transfer on hook". don't know if it can do all what i want. > > others: some china phones almost can do it, but need to press > > "hold" to cancel transfer. > > > > 3. call back on busy. in alcatel, if you dial someone and he is on > > the phone, you will hear something like "busy, please dial 5 if you > > want to request callback". you can dial 5 and you will hear "success, > > please hangup". asterisk has several ways and patches to do this. but > > i saw some phone can do this locally. i don't know which is better. > > linksys: has this function in spec. don't know how to use. > > snom: has "call completion". > > others: i didn't find this or i miss it. > > > > 4. pickup. i think this is easy to emulate "*8" and let asterisk do > > it. any better method? every phones can do this emulation. > > > > 5. three-way conference, forward. if there are simple (one key) method > > to implement these. in alcatel, if the phone if forwarded, when you > > pick up the handset you will hear like "forwarded, please press *1 to > > cancel". it's easy so everyone can cancel the forward. but it need two > > keys to start a forward, so few users know how to forward a number. > > > > please correct me if there are mistakes or missing. > > thanks again for your great help!! > > > > Regards, > > tbskyd > > > > > > _______________________________________________ > > --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