Thanks. However, I discovered a guide on doing this at the following url:- http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Example 2 shows to use a macro to present a menu to the member of staff before the call is bridged. Many thanks Dan -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy Sent: 17 March 2010 17:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Filtering We've done this with a bit of trickery - I believe we use System to copy a blank audio file into place before calling Dial so that Asterisk thinks the caller has already recorded their name. On 3/17/2010 1:42 PM, Dan Journo wrote: > Thats similar to how I want it to work, however I dont want the caller to > have to give their name (even the first time they call) > Is there any way of using the p option of the dial command, but totally > remove the caller name recording feature? > > Thanks > Dan > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy > Sent: 17 March 2010 15:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call Filtering > > Sounds like you want some type of call screening. Check out the p > option to the Dial command. > > >> Hi, >> >> I would like to develop a dialplan that allows the callee to reject >> the call like this:- >> >> 1) Call comes in and receives a greeting and get put into a queue. >> 2) A second call is placed to the member of staff (SIP phone or >> mobile phone) >> 3) The member of staff answers the call and is presented with a few >> options. >> 4) If the member of staff presses 1, the incoming call is connected >> to the member of staff. >> 5) If the member of staff hangs up or presses 2, the incoming call >> is sent to a voicemail box. >> >> The problem being, I can't see to place the second call without >> bridging the first call. >> >> Can anyone point me in the right direction? >> >> Many thanks >> Dan > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users