You should be able to use the Avaya IP Phone with CUCM using h.323 or SIP. However, you can use SIP endpoint only if you have CUCM 5.x or higher.
As for SRST, this might not be possible. 3rd party devices don't download the config file from TFTP like the Cisco phones do. It is in the confg file that SRST info is sent to the device. You will have to configure that on the SIP device manually if there is a feature in there. Also, you will have to configure a dial-peer and other SIP parameters on the SRST gateway so that the 3rd party sip device can fail over.
I have an IP Office 3.1.29 and CUCM 6.1.2 doing h323. Calls pass both directions, but no RTP is established and call goes busy after ten seconds going off hook. Any suggestions?
So if call setup works but audio is not setup means either you have a codec issue, as in both the end points are not able to negotiate an acceptable codec or h.245 negotiation is not completing.
You might want to take a look at the sniffer from the Avaya phone and confirm that h225 and h245 are completing fine.
Make sure you have
Direct IP-IP Audio Connections Y
IP Audio Hairpinning Y
turned off in your signaling group on the ACM.
I also frequently forget to fill in the
Trunk Group for Channel Selection
field in the Signaling group. I don't know why, but I have done this several times!