We are going to be migrating to a VOIP switch very soon and we have 2 Avaya Aura CM6 switches connected via PRI. One of the CM6's is on a different domain and different IP address. The question is what Media Modules are required for SIP trunking, and do I need a session manager? Both systems do not have a session manager and are not currently supporting sip phones. The phones on both switches will remain analog or digital, no VOIP. This is all on a campus environment.
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