Asterisk - Conferencing MulticastRTP and SIP channel together AFTER commands on run on the SIP channel

I have a PA system in the building that requires dialing an extension and then #00 to signify "All Call". Also, the phones can do pages through multicast.

I'd like to tie these two systems together.

Right now I have the following through AEL to join them both together

_XXXX => Page(SIP/${EXTEN}@announcementpbx&MulticastRTP/basic/224.0.1.116:9999,q);

I can pick up any SIP phone, dial 7145 (for pages). Asterisk will start the bridge and then I dial #00 to do an All Call.

The problem is, the multicast fires first, and also you can hear me dial #00 through the multicast. It's a weird combination.

What I'd like to have happen:

1. Dial 7145 from a SIP phone
2. Macro calls the extension 7145 (the PA system)
3. Macro dials (plays dtmf) #00 for me to do an all call
4. THEN put my original call, MulticastRTP, and the PA system into a conference. Mute MulticastRTP and the PA system so I can speak.

Does this make sense?