I am setting up a new SRST remote site and want to use a SIP trunk to the PSTN provider during SRST (this is their normal gateway). Are there any issues with this? Do I HAVE to use a traditional PSTN gateway (T1, E1 or Pots) during SRST?
I am using Callmanager 4.1 and will connect to the gateway during normal operations via H323 and then the ISR will convert to SIP to go out the gateway.
This should work. Your existing voip dial peers pointed to the SIP trunk will also be used during SRST. Just make sure that the existing destination pattern is what your users will expect to dial (9+number, for example).