I have a router that is setup to use SIP trunk with CUCM 11. Its all SIP phones. FXO is used for the PSTN and I have setup the connection plar to point to the AA extension.
My issue is when in SRST mode, how can I translate the connection plar to an extension on the SIP phone. The SIP phones automatically generate voip dial-peers so essentially I need to just find a way to translate the connection plar without disrupting the system when it is back online.