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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New to SIP trunks
- Thread starter jlroeder
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Theoretically you could terminate the sip trunks directly to CM if you have the licenses available. You want to execute the command display system-parameters customer options and on page 2 look for maximum administered SIP trunks. Normally though you want to have an ASBCE.
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I'd recommend talking to your business partner. I've done SIP trunks for my faxing solution and it took myself, the fax manufacturer, and my teleco to get it configured properly. You also need to make sure you have your QOS policy configured correctly. Honestly I'd look at Cisco because chances are it will be cheaper especially since Avaya won't be supporting 6.x in about 6 months.
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