I have one extension that is making calls automatically. I want to when call is connected, redirect that call to operator(another extension) on another extension. Is this possible ?
Thank you
In the phone dial plan try <:222>xxx This should redirect any number you dial to ex 222
The dial plan I use is ( P1 <:222>) . Now you only have to take the phone off hook. No key pad needed. It dial 222 after 1 sec off hook
Well i have 10 operators and when i am dialing i dont know which operator will be free. Ideally, i want to which extension it will be redirected, decided when the call is answered.
If you set up a virtual extension. In Follow-Me List all the operators. Set Ring Strategy to ringall. All Free operators will ring when you dial virtual extension number. The first operators that answer get the call
I dont think your advice is applicable to my scenario. I created Diagram to clarify my intentions. Well i have Extensions XXX that is auto-dialing some clients A and B(real mobile phone numbers) and when for example client A will pick up phone and call is connected i want that call to be redirected to one of my operators. I hope now you understand my scenario. I hope its possible to be configure it this way. Thank you for your help.
https://www.voip-info.org/wiki/view/Predictive+dialer
Look this simple example in github: https://github.com/navaismo/CallFile-Dialer or https://www.freepbx.org/xact-dialer/
Thank you for reply, but is it possible to configure it the way i wanted it ?
I know it can be done in etc/asterisk/extensions_custom.conf it’s will look something like “2001 => 2299; 6001,1,Dial(SIP/0,20)”. when ext 2001 dials 2299 they will be redirected to 0
exten => 0715551234,1,Dial(SIP/sip-phone,60)
When an incoming call reaches this point and matches extension number 0715551234, a sequence of two events is triggered, starting with the Dial() application, which connects together all of the various channel types in Asterisk. Here, it connects to a SIP channel, called sip-phone, which is represented by a section called [sip-phone] in sip.conf, with a ring-timeout of 60 seconds.