Problem with ocs 2007 not calling to internal. RRS feed

  • Question

  • Hi everyone,


    I’ve got a problem with the office communications server.

    I have a server installed with office communications server and I have another server where I installed the mediation server on. And I have a third server where asterisk is installed. And I have one extension and a real phone is connected to it.


    Now I’ve configured everything so that I can call from my telephone to the asterisk server and then the asterisk server route’s it to the mediation server and the mediation takes  the call to the communications server and the communications server takes it to the client. And that is working perfect without any problems at all. And I made an account and attached a number to it and made a route so I have to dial a 9 in front of the number so that asterisk takes it to the mediation server.


    Now the problem that I’m facing is that if I want to make a call from my client to the phone that it’s not working. I have logged in to my communicator client and i dial the extension number from my phone that is connected to the asterisk server. And then the only error I get is: the peer actively refused the connection. And I have a normalization rule that is saying that it has to route the number 202 to the mediation server and I think that is happening but I can’t really figure this problem out. And I found some information about this problem and I think that it has something to do with the certificate of the server. But I’m not really sure.


    And I have a log from a communicator client in windows 7 about this  error. You can see that I’m trying to dial the phone with 202 and then at the bottom it gives the error that the peer refused the connection but I don’t really understand it. So If anyone knows how to get this right of have any information for me so I can achieve my goal of calling the phone then it would be great. And this is how the log looks like :


    A SIP request made by Communicator failed in an unexpected manner (status code 80ef01f8). More information is contained in the following technical data:


    RequestUri: sip:202;phone-context=1@voip-test.nl;user=phone

    From: sip:edwin@voip-test.nl;tag=4d4e3a5001

    To: sip:202;phone-context=1@voip-test.nl;user=phone;tag=2EB7972C072CE377DA911CFE0D04FD16

    Call-ID: 39f84c36aa6a4294b64d50aa5791fcee

    Content-type: application/sdp;call-type=audiovideo




    Response Data:


    101 Progress Report

    ms-diagnostics: 14011;reason="Called Number translated";source="test-voip-server.voip-test.nl";RuleName="+3-Digit test";RuleDN="CN={6F49741A-4A7F-4B0E-A9B5-59C68611D7B0},CN=Location Normalization Rules,CN=RTC Service,CN=Services,CN=Configuration,DC=voip-test,DC=nl";CalledNumber="202";TranslatedNumber="202";appName="TranslationService"



    101 Progress Report

    ms-diagnostics: 12006;reason="Trying next hop";source="test-voip-server.voip-test.nl";PhoneUsage="CN={C491D082-9CD3-4A41-9A79-9DCEE38670EB},CN=Phone Route Usages,CN=RTC Service,CN=Services,CN=Configuration,DC=voip-test,DC=nl";PhoneRoute="Test-Asterisk";Gateway="server2008-medi.voip-test.nl:5061";appName="OutboundRouting"



    183 Progress Report

    ms-diagnostics: 12006;reason="Trying next hop";source="test-voip-server.voip-test.nl";PhoneUsage="CN={C491D082-9CD3-4A41-9A79-9DCEE38670EB},CN=Phone Route Usages,CN=RTC Service,CN=Services,CN=Configuration,DC=voip-test,DC=nl";PhoneRoute="To the PSTN";Gateway="Server2008-medi.voip-test.nl:5061";appName="OutboundRouting"



    504 Server time-out

    ms-diagnostics: 1007;reason="Temporarily cannot route";source="test-voip-server.voip-test.nl";ErrorType="Connect Attempt Failure";WinsockFailureDescription="The peer actively refused the connection attempt";WinsockFailureCode="10061(WSAECONNREFUSED)";Peer="Server2008-medi.voip-test.nl"



    and as I said before any help would be great ;D thanks in advance..

    Thursday, March 11, 2010 2:41 PM

All replies

  • Your asterisk server is "test-voip-server.voip-test.nl", correct? 

    Is it listening on 5060 (or 5061 for TLS?)

    The error "the peer actively refused the connection attempt" means the asterisk server is not listening or not accepting connections from that IP
    Mark King | C/D/H | MCTS:OCS | MCSE: Messaging | MCITP:Enterprise Administrator | CCNA
    Thursday, March 11, 2010 6:51 PM
  • By default Asterisk does not use SIP over TCP and only version 1.6 or greater supports SIP TCP, so worth checking that first. There is a good config walk through here http://blogs.technet.com/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx 
    Chris Clark - | MCTS:OCS | MCSE | MCSA | CCNA
    Thursday, March 11, 2010 10:53 PM
  • Thanks for the response :D

    my asterisk server is not "test-voip-server.voip-test.nl" that is my ocs server on my domain my server is named test-voip-server en my domain is called voip-test.nl i know it seems a little bit confusing but i hope you can understand me.

    and i configured my mediation server that is listing on port 5061 i believe for ocs and 5060 for my asterisk server and i know its working because i can call with a real phone to a communicator client. accept i can't go the other way the call from my communicator client to the phone doens't work and it gives me this error log.

    and i have asterisk 1.6 running and i did a enable tcp command but maybe i didn't do it right so maybe you could tell me exactly how i can make ik work with tcp.

    thanx for the tut btw but i had already seen that one but that doens't matter i love the input :) keep em coming and going :)
    Friday, March 12, 2010 11:09 AM