[HELP] Legacy PBX VoiceMail integration

I am looking at the possibility of replacing our legacy Rolm PhoneMail system with Asterisk VoiceMail. I am using an 8 port Intel PIMG (PBX IP Media Gateway) and a Cisco 3640 Router to interface betwen the Legacy and SIP environment. The PIMG provides the necessary signaling to the ROLM 9751 to provide MWI and stutter dial tone upon receipt of a MWI notification from Asterisk. There is an entry in sip.conf for the PIMG and for each legacy telephone that will use Asterisk for VoiceMail. The username field in the sip.conf entries are passed to the PIMG and used to signal the appropriate ROLM phone for MWI. All of this works great.

The problem is trying to get this to scale to 2000 mail boxes. The PIMG has 8 ports to interface to the 9751 and I will need at least 36 ports.

(1) How do I get asterisk to roll over (load share) to a second and third gateway when the 8 ports are all busy?

The Asterisk receives “-- Got SIP response 480 “Temporarily Unavailable” back from xxx.xxx.xxx.xxx” from the PIMG when all ports are in use. Instead of Queuing the Notification and trying again Asterisk just moves on to the next Notification. This creates the potential of having incorrect MWI states.

(2) Is there a way to have asterisk queue the MWI notification and try again until it receives a 200 OK message from the PIMG?

(3) Are the SIP responses stored in a variable in Asterisk?