Hi Mark I will try it, your solution (if it works)would be a standard feature in any implementation of Asterisk that I would do in the future. There is one small problem, no doubt you can see it. It's when the called extension doesnt answer and it goes to VM - the receptionist must hang up instantly so that the caller doesn't miss the first part of the VM message or if the caller doesn't want to leave a message at all. In the first case (goes to VM ) perhaps the default would be to reconnect the caller with reception before going to VM and then the reception asking if the caller wants to leave a message. Thus sending the call direct to VM (if he/she does). Of course one can always retrieve the caller on hold by pressing the flashing line button.
Thanks again Henry Henry Coleman [VoIP-PBX.ca] > Let me know if it does and how well if you could as I probably need to > implement it pretty soon as well, thanks, Mark. > ----- Original Message ----- > From: "Henry.Coleman" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Sunday, April 30, 2006 7:54 AM > Subject: Re: [on-asterisk] blind vs. announced transfer > > > Hi Mark, this solution sounds very practical. > I'll must try this out and see if it works. > > Thanks > > Henry > -- > > > Henry Coleman [VoIP-PBX.ca] > >> From the [EMAIL PROTECTED] handbook. >> >> >> Added to CVS HEAD (=Asterisk 1.2.0) in Jan. 2005: >> >> ;transferdigittimeout => 3 ; Number of seconds to wait between >> digits >> when transfering a call >> ;courtesytone = beep ; Sound file to play to the parked >> caller >> ; when someone dials a parked call >> ;xfersound = beep ; to indicate an attended transfer is >> complete >> ;xferfailsound = beeperr ; to indicate a failed transfer >> ;adsipark = yes ; if you want ADSI parking >> announcements >> ;pickupexten = *8 ; Configure the pickup extension. >> Default >> is *8 >> ;featuredigittimeout = 500 ; Max time (ms) between digits for >> ; feature activation. Default is 500 >> >> [featuremap] >> ;blindxfer => #1 ; Blind transfer, default is # >> ;disconnect => *0 ; Disconnect >> ;automon => *1 ; One Touch Record >> ;atxfer => *2 ; Attended transfer >> >> [applicationmap] >> ; don't use e.g #9 for applicationmap or featuremap unless you have >> changed >> 'blindxfer' from # to e.g. #1 ! >> testfeature => *9,callee,Playback,tt-monkeys ;Play tt-monkes to >> callee >> if >> *9 was pressed - use 'callee' or 'caller' >> >> If you set the variable __TRANSFER_CONTEXT, then that context will be >> used >> (note the two leading underscores). >> More on this: You need to set a TRANSFER_CONTEXT, either for the >> transferer >> or transferee channel. I dont know why, but res_features give priority >> to >> the transferee TRANSFER_CONTEXT, if not found, then use the transferer >> TRANSFER_CONTEXT. That context is used to match the extension to dial. >> So >> you can set this var to any context you want. >> >> Using the blindxfer in [featuremap] section you can redefine the >> transfer >> key. For example, if the blindxfer is set to "##", transfer only happens >> when you press the "#" key twice very quickly. This solves a problem >> using >> Asterisk phones to call IVR systems such as those used by banks and >> credit >> card companies - "Enter you account number followed by the # key". >> >> atxfer allows attended transfer or supervised transfer. It works like >> this: >> >> While on conversation with another party, you dial the atxfer key >> sequence. >> Asterisk says "Transfer" then gives you a dial tone, while put the other >> party on hold music. You dial the transferee number and talk with the >> transferee to introduce the call, then you can hang up and the other >> party >> will be connected with the transferee. In case the transferee does not >> want >> to answer the call, he/she simply hangs up and you will be back to your >> original conversation. >> >> Note: You MUST use the T and/or t options in the command Dial() in order >> to >> allow the caller and/or callee to use any transfer feature >> >> ----- Original Message ----- >> From: "Nabeel Jafferali" <[EMAIL PROTECTED]> >> To: <[email protected]> >> Sent: Saturday, April 29, 2006 7:40 PM >> Subject: RE: [on-asterisk] blind vs. announced transfer >> >> >> Henry: >> >> Blind Transfer and Attended Transfer are both provided for in the SIP >> standard, and also in Asterisk's implementation of SIP. It is up to the >> phone. >> >> The Cisco 7960 that I use regularly has both options. So does the snom >> 320 >> currently on the desk in my lab. I'm sure other phones do to, the exact >> ones >> I'm not sure of right now. >> >> Also, the Cisco and the snom can also interrupt the Attended Transfer >> and >> "take back" the call. >> >> Nabeel >> >>> -----Original Message----- >>> From: Apache [mailto:[EMAIL PROTECTED] On >>> Behalf Of Henry.Coleman >>> Sent: April 29, 2006 6:44 PM >>> To: [email protected] >>> Subject: [on-asterisk] blind vs. announced transfer >>> >>> I have a client who would be very happy with * but for one thing. >>> Actually its not asterisk per-sec, it's that he used to have >>> a key system and * is a PBX. For those who enjoy a challenge >>> here it is: >>> >>> A PBX is superb at the blind transfer of calls: >>> Answer the ringing line hit the TRANSFER button, dial the >>> extension number and hit SEND and the call is off the "board" >>> and it's on to the next call. >>> >>> The problem comes when you need to "Announce" the call to the >>> extension before you send it. On a Key System the attendant >>> can refer to incoming call as being on "Line (x)" and the >>> person can simply select "Line (x)" >>> Obviously with a pbx you can't do this. >>> >>> The challenge then is to be able to announce and transfer a >>> call in one step. This would add a significant feature to *. >>> As far as I am aware no PBX can do this without using a two >>> step process. Here is how I think it should work- >>> >>> Answer the ringing line, >>> select TRANSFER, dial the extension number, announce the call >>> and press the SEND button connecting the incoming call with >>> the extension. Sounds like the first definition but is light >>> years different in functionality. >>> Before you go ahead and solve this don't forget that >>> sometimes the person at the extension will say "no I don't >>> want to talk to this caller" so there must be a way to send >>> the call to VM or reconnect to the caller. >>> >>> >>> I'm using 12 x GXP 2000's in this system Please substitute >>> the codes for "TRANSFER" and "SEND" >>> >>> The customery beer at Tobys awaits the first person to solve this. >>> wait!... make that two beers. >>> >>> Henry >>> >>> -- >>> Henry Coleman [VoIP-PBX.ca] >>> >>> -- >>> Henry Coleman [VoIP-PBX.ca] >>> >>> >>> >>> --------------------------------------------------------------------- >>> To unsubscribe, e-mail: [EMAIL PROTECTED] >>> For additional commands, e-mail: [EMAIL PROTECTED] >>> >>> >>> >> >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [EMAIL PROTECTED] >> For additional commands, e-mail: [EMAIL PROTECTED] >> >> >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [EMAIL PROTECTED] >> For additional commands, e-mail: [EMAIL PROTECTED] >> >> >> > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > > >
