RTP Receiver Error on LAN/Exchange 2007 UM

Discussion in '3CX Phone System - General' started by jingram, May 1, 2008.

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  1. jingram

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    OK, I did a search and wasn't able to find anything specifically related to my scenario. At this juncture we have 3cx up and running, a couple of grandstream GXP 2000s registered and are able to call to each other via extension with no problems. We are testing the exchange 2007 demo license and have followed the directions setting up 3cx and Exchange 2007 for UM. All we are attempting to do at this juncture is dial an extension and test the automated attendent and voicemail capability. The issue is that we never hear the automated attendant. We hang up the call and we get the emails coming in from exchange with no issues. Our logs on 3cx show this:

    14:30:29.398 MediaServerReporting::RTPReceiver [MS105000] C:7.2: No RTP packets were received:remoteAddr=192.168.x.x:38400,extAddr=x.x.x.x:10970,localAddr=x.x.x.x:10970
    14:30:29.304 Call::Terminate [CM503008]: Call(7): Call is terminated
    14:30:29.304 Call::Terminate [CM503008]: Call(7): Call is terminated
    14:30:17.898 CallCtrl::eek:nLegConnected [CM503007]: Call(7): Device joined: sip:100@exchange.x.com:5060;transport=TCP
    14:30:17.898 CallCtrl::eek:nLegConnected [CM503007]: Call(7): Device joined: sip:100@192.168.x.x:5060
    14:30:16.867 CallCtrl::eek:nRerouteReq [CM503005]: Call(7): Forwarding: DirSip:RecordMessage@[Dev:sip:MSExchange@exchange.x.com:5060;transport=TCP]

    Where the RTP Error is, the external address and the local address are both the external facing IP of the exchange server, which I found a little odd.

    This occurs when call missed/call busy options are set to forward to voicemail. When forward to autoattendant is chosen the phone shows a 480 error.

    This is stumping me... would love some help!

    Rgds,
    Jack
     
  2. jingram

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    Anybody? I would love some insight here. I can't imagine that I am the only one who has ran into something like this.
     
  3. jingram

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    Well, it ended up being a firewall issue. Since we were only doing internal testing, STUN wasn't needed. Since we use our firewall as our default gateway, all RTP packets were trying to route out to 3CX for STUN which was causing us a major headache since the firewall was dropping those unhandled packets. Turned off STUN in the PBX and everything started working immediately, although 3CX really needs a simple checkbox to turn STUN on and off to get the service to stop reporting itself in the server logs every 15 seconds!

    Jack
     
  4. Nick Galea

    Nick Galea Site Admin

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    Hi,

    However note that without STUN outbound calls via a VOIP provider or to external extensions wont work. Yes an option to switch of STUN will be considered for future versions.
     
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  5. frank2059

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    Hi,

    I've got the same problem in my "home lab".
    When will this feature be available?

    Regards,

    Frank
     
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