----- Original Message ----- From: "Shawn L. Djernes" <[EMAIL PROTECTED]> To: "'Asterisk Developers Mailing List'" <asterisk-dev@lists.digium.com> Sent: Tuesday, February 22, 2005 10:52 PM Subject: RE: [Asterisk-Dev] Refined Voice CallerID Announce
> What about those of us who want the features and sound quality of a SIP or > IAX phone but can not see the display to read the caller ID? I have been > looking for a way to have the Caller ID played to me before I get dropped > someone I may not want to talk to. But have yet to see a way to get any app > but meatme to bridge the two channels of in progress calls. > > I think the example shown should work but how do you bridge in the calling > party after hearing the caller ID info. > > Shawn Exactly my point I want to put out there. Some people actually besides me want this feature. I think a drop call feature should be put into it to, keep the party ringing while the information is played and if you want to talk to the person you let it finish and wrap up until a connection is made If you dont want to talk with the person have like the "#" button used to transfer them down the dialplan so if voicemail is next then they go to voicemail, otherwise whatever is specified Layout 1. Caller Calls in 2. Caller ID Collected 3. Asterisk Answers 4. Call Directly Rings to a phone -or- is transferred using a menu 5. Party's Phone Rings and Caller gets ringing situation 6. Party Picks up phone hears caller ID Information either continues the call or "#"'s it to the next dialplan rule 7. Everything goes well Now how do we go about implementing a Vocal Caller ID could name it app_vcid.c I think festival would have to be used and it would have to control different subsections of asterisk and how they interact. I.E. Keeping the calling party ringing until the CID is played thru and ends, then stopping the ringing for the caller and bridge the phones together or is interruped by a "#" selection by the answering phone and is sent to Voicemail or whatever the next step is in the dialplan for that extension. So basically it would be like the Dial command but maybe VDial just some enhancements or it could just be simply intergrated into the Dial Command and have it set as an option for each SIP/IAX phone you want to use it on under thier options VCID=1/0 My only worry about this is how much process time this would eat up on a box with 80+ concurrent calls happening at the same time but maybe will soon find out? _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
