Nortel CS 1000 and Speech Server 2007 RRS feed

  • Question

  • I am trying to help facilitate a Speech Server 2007 configuration as a test in our company. I have been developing on Speech Server for about 4 weeks after discovering it(Wow!). I am using a soft phone right now to test but I want to setup a test using our regular phones(test for my Management). We have a Nortel CS 1000 (Software Release 5.0)configured with VOIP(already using VOIP for some of our phones). From several blogs and message boards I have gathered a little bit of information on how to configure our Nortle CS 1000 (I am a .NET developer so I am stupid on the Nortel devices).

    Here is what I have found anyway:


    1) Configure the Speech Server 2007 as an End Point on the Nortel CS 1000. (Our support provider(not Nortel)  for Nortel did not unhderstand what I had found so I am not sure if this is correct.) (How do we do this if this is correct?)


    2) Confiigure the route in the  NRS to route the call to the Speech Server. (Not sure how this is done either.) Our provider didn't know either Smile


    It does not apopear that we need any thing else but I am not sure.


    Any advice would be greatly appreciated.



    Senior .NET Developer

    Thursday, March 20, 2008 12:31 PM

All replies




    Haven't set up Speech Server at all yet, but as far as the CS1000 Rel 5.0 is concerned, it supports Gateway Endpoints and User Endpoints.

    • A Gateway Endpoint is a PBX-type system (or OCS2007 server etc.) that has 4-digit Coordinated Dialing Plan (CDP) entries and 3-digit Uniform Dialing Plan (UDP) location codes associated with it, depending on your dial plan and the number of digits in your extensions and location codes of course
    • A User Endpoint is a single SIP UA that has a single 7-digit location code + extension number associated with it


    Both are set up under the Network Routing Service (NRS), a separate application that runs on Nortel's Signalling Servers.  Switch to the Standby database to make changes, and go to Configuration -> Gateway Endpoint (or User Endpoint) to create this on the Nortel side.  Note that if you are creating a Gateway Endpoint, the Routing Entry (location code) is created on a separate screen after creating the Gateway Endpoint.
    Monday, March 24, 2008 6:11 PM
  • Thanks so much for the info. This will really help get this project moving.




    Tuesday, March 25, 2008 12:04 PM
  • Here is the latest info. I guess that we are suppose to go with the Gateway endpoint since that is the only one that we can enter a IP address into. We configured the Gateway endpoint and set the route of a specific extension to that Gateway endpoint.  We then activated the Database and proceeded to make a call. We received a busy signal. I put a sniffer app(Wireshark) on our Test Speech Server 2007 to monitor traffic coming in and when we dialed the extension we saw nothing from the signaling server or any other nortel server. We did not see any SIP traffic coming in at all.  I tested the sniffer app with my softphone on my PC and I was able to see all of the SIP traffic from my PC. It seems that the NRS is not routing anything out to the speech server.


    Is there anything else we need to look at?


    Is it possible to setup a User Endpoint with an IP Address? We cannot seem to find that field for the User Endpoint.



    Wednesday, March 26, 2008 7:23 PM

    We are working through the ‘exact’ same issue.

    I have a ticket open with MS right now and two techs are working on it.

    I sent Speech Server logs, Event Logs, and Wireshark traffic captures.


    We are able to get some SIP traffic but it fails in a fast busy tone.


    The Server is set up to accept any incoming calls ‘*’


    The CS1000 is set up to use TCP only,

    And the G.711 codecs Only.


    As soon as I get an answer I will post…

    Thursday, March 27, 2008 7:01 PM
  • Thanks Mike.  Hopefully you can get some info on this. Yeah, we cannot even see SIP traffic coming from our CS 1000 to the Speech Server. I was on the phone with our Nortel Tech today and they are not too familiar with the Speech Server portion. He seemed to think there was something wrong with our NRS route since we could not see any traffic. He was actually going to try and setup a Speech Server possibly to investigate.  He was pretty sure we needed to setup a Gateway Endpoint for Speech Server though. That at least answered my Endpoint question. Now we will try to figure out the NRS routing to see what is going on.


    Thanks again.





    Thursday, March 27, 2008 8:27 PM
  • Mike,


    Since you are seeing SIP traffic from the CS 1000 you are farther ahead than we are. Here is what we have done to configure an extension to go to the speech server. Maybe you can offer some advice on this.

    1) Created a Gateway Endpoint

    IP Address 192.168.x.x

    SIP Support: Static SIP endpoint

    SIP Transport: TCP

    Sip Port: 5060


    2) Create Route for extension

    Gateway endpoint: Chose the one created above

    DN: 8900 extension we wanted to test with

    CDP Steering code

    SIP URI cdp.udp


    Is there anything else we sould do?




    Friday, March 28, 2008 12:09 PM
  • SIP URI cdp.udp?


    Is there a cdp.tcp option?

    Friday, March 28, 2008 1:12 PM

    no cdp.tcp. I think this stands for something else on the CS 1000. I think it is a dialing plan or something. Nortel sent me some screenshots of their Exchange Server and OCS configurations and that's what it had too.  The gateway endpoint is where the protocol and transport are set. Never thought I would understand any of this until I had all of these issues. Smile



    Friday, March 28, 2008 1:44 PM
  •  ScottSwann wrote:


    Never thought I would understand any of this until I had all of these issues.




    It's amazing what you can learn when you are forced to do it the hard way. Keep us informed.

    Friday, March 28, 2008 2:43 PM

    I am the Nortel Engineer working with Scott on this. 


    I have the CS1000 Forwarding calls to the Speech Server currently, but the Speech Server is rejecting my call with the following error:


    Code Snippet

    The Telephony Manager declined a call with Call Id '19a7b758-3d688e2f-13c4-40030-5009ab-4bc2319d-5009ab' for the following reason in component telephony session: 'The media description received from the remote SIP peer has an invalid content type 'multipart/mixed'.'. 
    Further trace information for support personnel follows:
    Microsoft.SpeechServer.Core.InvalidMediaException: The media description received from the remote SIP peer has an invalid content type 'multipart/mixed'.
       at Microsoft.SpeechServer.Core.MediaNegotiation..ctor(LoggingContext loggingContext, ContentDescription rtcRemoteMediaOffer)
       at Microsoft.SpeechServer.Core.TelephonySessionInbound.CreateMediaNegotiation(ContentDescription rtcMediaDescription, SessionInfo sessionInfo, CallInfo callInfo, IPEndPoint sipPeerEndpoint)
       at Microsoft.SpeechServer.Core.TelephonySessionInbound.Initialize(SessionInfo sessionInfo, SessionReceivedEventArgs e, CallInfo callInfo, EventSerializer serializer, SpeechSession speechSession, Boolean isTlsConnection)
       at Microsoft.SpeechServer.Core.TelephonySessionInbound..ctor(SessionInfo sessionInfo, SessionReceivedEventArgs e, CallInfo callInfo)
       at Microsoft.SpeechServer.Core.TelephonyManager.CreateSession(Int32 inviteReceivedTickCount, SessionReceivedEventArgs e)
       at Microsoft.SpeechServer.Core.TelephonyManager.SignalingSessionReceived(Object sender, SessionReceivedEventArgs e)



    I am using the same Trunk configuration that I have for integration of the CS1000 to both Exchange 2007 and to OCS 2007.  Both of which have no issues. 


    My Speech Server is configured as:




    I've not spent any time with Speech Server until Scott asked for some assistance so I'm not familiar with the platform and it seems to be running a different engine than the current OCS.


    Maybe someone on the forum here will have some knowledge as to what Speech Server is expecting from our side to get this working.  I've read it looks for a standard SIP message, but that can mean a whole lot of things.  I've searched online and there is nothing that documents the Warning that I get on microsofts site that I can find, and the link in the warning goes to a page at microsoft that says there is no more information available on this particular event.


    Code Snippet

    Source: Office Communications Server 2007 Speech Server

    Category: Telephony Application Host

    Event ID: 32768





    For anyone trying to get to this point from the Nortel here is the basic configuration that you will do to get to this point (Assuming you alreadly have a working NRS):

    1) on the CS1000 create a route in your Dial plan that will forward calls to a SIP Trunk

    2) Create a SIP Trunk that will route up to your NRS

    3) Create a Static SIP Endpoint in NRS that points to the Speech Server

    4) Create a Destination Route that maps the extentions you made in Step One ot the Endpoint that you created in Step 3.

    5) Create the Trusted SIP Peer in Speech Server as a reference to the Node Address of the Signaling Server (If you don't know this address your in luck as an easy way to find out the address is to look in your event viewer and it will give you the following error with the address whenever you try to make calls)

    Code Snippet

    The Telephony Application Proxy declined a call with Call Id '19a79368-3d688e2f-13c4-40030-50054d-27317ec2-50054d' from address 'AA.BB.CC.DD' because the remote address is not authorized. Only calls from peers in the trusted SIP peer list or the debugger on the local host are allowed.


    The AA.BB.CC.DD in this error will be the address that you will want to use.
    Hope this post is helpful, and hopefully someone on the Speech server will know what it is looking for so that we can get this working for everyone out there
    Monday, April 7, 2008 2:51 PM
  •  Marshall Harrison [OCS MVP] wrote:

    SIP URI cdp.udp?


    Is there a cdp.tcp option?


    CDP.UDP in the URI is not in reference to UDP vs TCP traffic


    CDP -> corresponds to Coordinated Dial Plan (this is going to be your 4/5/6 digit dialing where you know my number is xxx6748 you can just dial 6748 to reach me when we are on the same CS)

    UDP -> corresponds to Uniform Dial Plan (this will be your ESN dialing typically where you will have to dial my full number)


    It's all apart of the NARS/BARS.


    In this case Scotts dial plan is a CDP as it is CDP.UDP.  Think of this as you would with networking and containers where it's storing it's path to get to the device.  It stacks them on top as it goes further down the structure.

    Monday, April 7, 2008 3:02 PM
  • It would sure simplfy things if companies would quit reusing acronyms for different things :-)


    Monday, April 7, 2008 4:46 PM

    Sorry for the long delay.

    The MS techs we were working with (Ticket SRX080312600668 if any of you MS guys want to look it up)

    sent back this as the last useful communication:



    I looked at the event and ETL logs and found that the Content-Type is not set to the value that MSS expects. The working call in one of the test machine sends Content-Type:multipart/sdp



    The Telephony Manager declined a call with Call Id '1ee41fa0-11c7a8c0-13c4-40030-b74d0-4c8d3527-b74d0' for the following reason in component telephony session: 'The media description received from the remote SIP peer has an invalid content type 'multipart/mixed'.




    I have been the middle man between MS and Nortel this whole time so I forwarded that onto Nortel and after a few days I was sent this note:


    >>Mike…The SIP problem has been turned over to the ICA Team.

    >>ICA works specifically with the Nortel and Microsoft integrations.



    Just to recap for everyone else just joining:

    I have a Windows 2003 server, running OCS 2007 Speech Server eddition.

    I have developed an employee information tool on it using Visual Studio 2005

    with Speech SDK and Foundations for Workflows.


    Everything works great in Debug mode using the Simulator from my desktop.

    Everything works great using "X-Lite SoftPhone" from 3 different desktops.

    (Dial String for X-Light Soft Phone: "sip:test@SpeechServer:5060;transport=tcp"

    (I should point out that I did not make any OUTGOING calls to the Xlite Phone because I didn't have a server to register it.)


    Our first attempt to interface the Nortel CS1000 and the MS Speech Server was through a 'Linux ASTRISK box using a UNISTIM patch.  I think the Unistim patch was working but the SIP communications kept failing.  That is what lead us to the software upgrate V5.0 for the CS1000.


    I do not know everthing that was done software(firmware) wise on the equipment but the hardware changes

    include the removal of our old IP phone card ( NTVQ5511/RLSE10) 

    Then into each of the 4 cabinets that make up the system a (MGCNTDW60BBE5) card was added, these each have 4 Ethernet connections and seem to tie the cabinets together.  This was done with some fiber cards.  Also a new card ( the cpu ??)  numbered (NTDW61BAE5 RLSE03) was added to the first cabinet.

    A high speed switch was also mounted by the phone equipment that all 4 of the '..BBE5' cards connect to (as well as each other)  I can't tell but it looks like some sort of MESH network.

    For reference: I have several (over 40) phones in the plant; most are analog, about 1/3 are digital sets and about 25 are IP based. (Not SIP-IP but Nortel UNISTIM-IP)


    While I was waiting for the upgrade to finish I installed a Diva Server V-Analog-4P (4 port analog voice modem thing) in the Server 2003/ Speech Server box and it worked great.  It came with 4 ports but you need to download the free liscense from the website to activate them.. It's a detail that caused a headache.. But easy to fix now that you know about it.


    The problem is that the DIVA can only handle 4 ports and it was just for devlopment testing anyway.

    Our true goal is to get SIP running to tie the server directly into the CS1000.


    Once  we had our CS1000 caught up on firmware to V5.0 and finished our CallPiloit voice mail install we went back to getting SIP (or goal all along) working.




    I did try to make the LINKSYS SPA3102 (Voice Gateway) unit work but it has having trouble with the "RE-INVITE"  I am assuming for now that my lack of telephony knowledge was the limiting factor on that failed experiment.  If anyone knows the settings I need I'd be glad to try it again and report back.


    If anyone wants to know any specific settings I have enabled on the CS1000 or the Server just ask..

    As I said before, as soon as I get an answer (apperently from Nortels ICA team now) I will post it.


    Good luck to all..


    (PS.  after I finish this headache My next one is that I have been asked to inteface the SPeech server with our fleet of Nextel/Sprint Push to talk cellphones.  We have about 80 onsite and they are the primary mode of communication for our employees. 

    I think I can get the Speaker and Microphone interfaced through a simple sound card Or maybe thorugh that SPA3102.  (I'm an Electronics Engineer by training, I'd just modify the audio signal and feed it into the box, It would think it's plugged into a analog phone set)

    The challenge is the DIALING OUT.  I would need a way to send the key-presses from the speech server to the connected Phone. 

    Also it would be nice if I could capture the display data on the phone so I could get a CALLER ID type of input when someone calls the system.


    Wondering what that would be like?

    Think back to star-trek.. Kirk flips open his "communicator" and says "Computer."

    Suddently you hear "~working~" as a response.


    Yep.  No rest for the Tech-Wicked.


    -Michael Ward




    Saturday, April 26, 2008 6:38 PM
  • Monday Morning update:

         Issue has been passed to a Lead Developer.
         I was told that the connection DOES work

         and IS supported.  

        Originally Nortel said they did not support

        the MS "version" of SIP (???)  It was the MULTI-PART

        /MIXED content that was the issue.  Now it appears

        that nortel does support it however I have not

        been told how. 



          I was told that a gateway was needed once.

          the gateway converts the SIP autoformat

          (Usually ulaw) into microsofts .mwv format.

         This step IS needed for Outlook and Office

         applications as they need the audio format changed.


         For this project, both the Server (MSS2007) and the

         Nortel system (CS1000) are configured to use  ulaw

         as the only audio codec.  Thus no gateway should

         be needed.


    And the saga continues...




    Monday, April 28, 2008 2:14 PM
  • Thanks Mike for the Updates. We are still strugling with this also. I have had to focus on some more important issues for the last few weeks so I have not had much more time on this either. It seems we are stuck at the same point you are. Hopefully they(Nortel team) can resolve this soon. I am curious to see what is the solution.


    Tuesday, April 29, 2008 2:05 PM
  • I thought I would post some more information I received from our Nortel Contact. He has been working on getting the Speech Server 2007 working in his lab. He was finally successful at getting it to work. He informed me that Speech Server 2007 is not completely supported on the CS 1000 as of yet. He was able to get it working by installing the Nortel Software release 5.5, which resolved some Exchange Auto Attendant issues. He was able to configure the Speech Server 2007 dial number as a Exchange Server Auto Attendant number and the SIP messages were then formatted appropriately for Speech Server to understand.  We have not had time to work on this internally yet. I would still like to attempt to setup this as a test to see if there any other issues. 


    If I discover anymore answers I will post them here.





    Monday, May 5, 2008 7:43 PM
  • Eureka!


                Install Nortel patches

                            MPLR 24399

                            MPLR 25226


                Set number  type from CDP to INTERNATIONAL


    Suddenly I have Nortel talking to my Speech server.


    I will try to narrow down any other settings but I think these three

    Were the ones that solved it for me.




    Thursday, May 15, 2008 8:51 PM
  • Michael,


    Tell me more!

    1) What version of the CS 1000 OS are you running 5.0? 5.5?

    2) How was the Speech Server setup on the CS 1000?  Did you configure it as a Static Gateway Endpoint or something else?

    3) So the extension you created was configured to use INTERNATIONAL instead of CDP.


    Thanks for all of the info you have placed here. I look forward to your reply. I really would like to get this setup at my company.



    Thursday, May 15, 2008 9:11 PM
  • Michael,


    If you get a Chance Michael send me an email on this. I would like to ask some more questions on this.


    Scott Swann



    Friday, May 16, 2008 8:05 PM

    Hi Micheal,


    I am new to speech server.


    I have installed speech server 2007.


    I could call speech server from softphone.Now i am trying to call speech server from one of the work phones. The problem here is,my phone recognises speech server port and establishes INVITE but my speech server doesnt recognise my phone port,it says port unreachable. My phone is Allworx handset(Model Number:9112).


    I really appreciate if you can let me know if i am missing anythign here and what should i do for my speech server machine to recognise handset's port.


    1128 1020.779165 SIP/SDP Request: INVITE sip:1234@, with session description

    1129 1020.779206 ICMP Destination unreachable (Port unreachable)

    1130 1020.968144 Allworx_02:01:1b Broadcast ARP Who has  Tell

    1131 1020.968720 UDP Source port: ip-blf  Destination port: ip-blf

    1132 1021.404892 UDP Source port: ip-blf  Destination port: ip-blf

    1133 1022.170139 Allworx_02:01:1b Broadcast ARP Who has  Tell

    1134 1023.526777 Allworx_02:01:1b Broadcast ARP Who has  Tell

    1135 1023.527315 UDP Source port: ip-blf  Destination port: ip-blf

    1136 1023.678026 UDP Source port: ip-blf  Destination port: ip-blf

    1137 1023.769310 SIP Request: CANCEL sip:1234@

    1138 1023.769340 ICMP Destination unreachable (Port unreachable)



    Thanks a lot in advance.
    Monday, July 14, 2008 9:35 PM
  • ;') FYI the telephone guys used UDP first. So good luck :')

    Thursday, September 25, 2008 9:25 PM
  • We have a 3Com VCX SIP phone system 'mostly' working with MS UM. I say mostly as we are still in the process of writing our own MWI software.


    Taking a quick look at your post, you've set the endpoint to be TCP port 5060.


    I believe the speech server is listening on port 5065 by default (edit: im wrong, the um worker process is on 5065 / 5066, main SIP is 5060).


    Sorry if I'm telling you something you already know.

    Wednesday, October 1, 2008 6:01 PM