asterisk expert needed cli function to add call leg rev3

Using asterisk 12.6.0 / FreePBX 12.0.1rc29 I will not be able to provide server access. You must use your own server for testing. Part 1: I need a method to add a leg to a call through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script. For example, 1231231212 from the outside calls inward, it rings through a ring group, and ext 500 picks up. I want to have a button on my crm call a php script to transfer the call that is ongoing between the outside number and ext 500, to another internal ext 100, or even external phone number. By the end of the function being completed, there will be a joined called from ext 500, the external number, and ext 100. The user at ext 500 will end the call by hanging up, while the call between the external number and the ext 100 continues. In some types of failure, a graceful error would be nice. The second internal extension should get the original caller id information passed through as is. I will handle the crm integration. When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address. If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui. Part 2: I need a method to transfer a call to a wav file through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script. For example, extension 123 internally, calls an external number. 123 starts the call, and is tranfered to the remove voicemail. 123 would click on a button that transfer the call to a wav file, which is played, and then the call is ended. In some types of failure, a graceful error would be nice. I will handle the crm integration. I will provide properly formatted wav files. When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address. If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui.