Configuring Cisco Phone System

We received the following email from our SIP Trunk provider:

With the Cisco Call Manager w/ CUBE, the CUBE is not passing UPATE messages to the Call manager. Thus, the CUCM will disconnect the call after 30 minutes, since no UPDATE was received. The CUBE must remove the UPDATE in the "Allow-Header" field.

This command will remove the UPDATE from any request or response message type:

voice class sip-profiles 1
request ANY sip-header Allow-Header modify ", UPDATE" "" 
response ANY sip-header Allow-Header modify ", UPDATE" "" 

The "voice-class sip profiles 1" must be applied to the dial peer facing PAETEC.  In this example we are using dial-peer 9000

dial-peer voice 9000 voip
destination-pattern .T
voice-class codec 1
voice-class sip localhost  dns:astrial.pe.mcleodusa.net
voice-class sip outbound-proxy ipv4:xx.xx.xx.xx
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2

So here are my questions:
1) We need to make these changes in the SRST device which is a which is a Cisco 2921/K9, correct?
2) The first part looks pretty straightforward, but what about the second set of instructions? I guess I need the Dial Peer explained to me and which IP address is supposed to be in the outbound proxy.
JesusFreak42Asked:
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agonza07Commented:
How many routers do you have in place? SRST is not neccesarily the CUBE, although it usually is.

On the dial peers, just find the one with the "session protocol sipv2" pointing to your provider. There's probably not many, so if you want to sanitize and post I can tell you which one to use.

Probably best to sanitize and just post your config....
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JesusFreak42Author Commented:
Does sanitize mean make private?
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agonza07Commented:
Just clean up your public IP addresses, remove usernames and passwords, etc. That way you dont post any identifying info online.
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