Well today I undertook the setup challenge of performing an ITSP setup on one of my sipXecs (Test & Tears) servers. I must say the experience was a bit challenging at first with just plain trying to figure out the process. I have listed a link below that was a little help in the process. The setup of the SIP Trunk Gateway was straight forward and the setup of the VoIP.ms account was a little stressful. The most challenging part of the process was configuring the 1:1 NAT or Full Clone NAT on the WatchGuard firewall. In the end it always comes down to the simply statement of "Oh Now I See"! I was in the end (2 -3 Hours) able to get things configured and working so that the SIP SBC Trunk Gateway is listed as "Authenticated" under the Diagnostics Tab - SIP SBC Trunk Gateway area. While my testing thus far has gone very well I am in no way ready to go "Only" ITSP and lay aside my assurances with traditional SIP gateways connected to PRI or POTS circuits. I am starting to believe though. Areas Tested: Outbound Dialing: Good Results Thus Far Inbound Call Routing from VoIP.MS: Good Results Thus Far VoIP.MS DID routing to sipXecs Accounts: Good Results Thus Far MOH Testing: Good Results Thus Far The only problem that I sometimes have is when I first dial a number. The softphone shows that the call failed however my cell rings a couple of times. When I try to do this a second time the issue does not occur again. If I wait a few minutes and try again the issue does not show up. If I wait a long while then the problems occurs again. I am not sure if the gateway connection to voip.ms is interrupted if no calls are made for a while. My thoughts are Session Timer Interval or Registration Interval in the SIP Trunk Gateway configuration. Perhaps it could even be a Firewall Issue. Thoughts anyone! I will be posting later a simple setup guide for (VoIP.ms) with sipXecs. Those that are more informed, experienced and intelligent than I, please be gentle. I would like to in the end have a good instruction set for the user community when connecting to VoIP.ms. The suggestions and corrections made by those that are far smarter than I can only be considered a greater asset. http://blog.myitdepartment.net/?p=191 Thanks Everyone, Rob
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