I have a callcenter customer in my asterisk.
They are calling from a CRM system with a build in SIP phone.
It is just a SIP client like X-lite.
But they want the call to be put to their SIP phone.
This is because the poor computer does not handle the audio streaming.
They don't want to push the number on the telephone, but click the number on the screen, but want to talk in the telephone.
let us say they have ext 100 on the sip client on the screen, and 101 ond the voip/sipp telephone.
when the 100 is making an outboud call, the 101 starts to ring, and when they lift the handset, they will hear the outgoing call.
I need a outbound route, or a context that I can put on the extensions.
I am using asterisk 1.8.5.0 with Freepbx 2.9.0.7
It a live system, and can't be broken during business hours.