Trying to connect a Siemens optiPoint 420 Standard S SIP Phone to OCS 2007 R2 Enterprise Consildated Environment RRS feed

  • Question

  • I may be going out on a limb here but I'm trying to accomplish just what the subject says.  In order to maintain our investment in our Siemens phones, we need are wanting to convert our Siemens 420 IP phones to SIP and connect to OCS 2007R2.  We've got the SIP firmware installed but we're lost on how to configure it.

    The SIP configuration screen asks for the follow:

    SIP Routing: Gateway or Server
    Registrar IP address or DNS name & Port
    Server IP address or DNS name & Port
    Gateway IP address or DNS name & Port
    SIP port
    RTP Base port

    Outbound proxy: Yes or No
    Default OBP domain name
    SIP transport: TLS, UDP, TCP
    SIP server type: HiPath 8000, Broadsoft, Sylantro, Other
    SIP session timer enabled: Yes or No
    SIP session timer value: expecting seconds
    Registration timer value: expecting seconds
    SIP realm
    SIP user ID
    New SIP password
    Confirm SIP password
    Transaction timer: expecting ms
    Registration backoff timer: expecting seconds

    With the SIP Routing configured as Gateway and with the registrar, server ip and gateway set to the IP Address of my Mediation server, I'm able to get a dialtone but nothing more.  If I attempt to dial the phone reboots.  It also doesn't receive a ring when someone calls.

    Anyone have any ideas I can try?

    Friday, February 20, 2009 6:41 PM

All replies

  • This is not supported. OCS uses the RTAUDIO Codec which is NOT supported by any standard SIP devices. Only phone certified for OCS will work (such as Polycom CX-700). Sorry.

    Friday, February 20, 2009 11:09 PM
  • Hello Grifter, Hello Mark,

    @Mark you are correct regarding the Support and Certification, but that no other Open Standard SIP phone will work is wrong. BTW: Status of Poly CX-700 and LG8540 is not certified! It's is called "optimized for Microsoft Communicator". No word about the Server :)

    Maybe your not familiar with the Snom OCS Edition - standard IP-Phones! For a lot of Details + Links Please check out this PDF document

    @Grifter: trying to register at mediation server will never work, cause it is no registration. What you have succesfully done, was sending an invite to Meditions Server. This is possible cause you dont need SIPoverTLS or NTLM/Kerberos Auth. from outside the OCS-Network. You played like a VoIP-Gateway on your Siemens.

    What you need are phones that can nativly register at OCS Edge Server, Standard Front End, Director Role and or Enterprise FE Server!

    The challenges here are not the SIP-Registration, which is pretty Standard-SIP. The big pitfalls are SIPoverTLS, NTLM and Rosenberg's Draft-based ICEv19 + Early Media. I can ensure you, you will never see this on a Siemens. To support Microsoft OCS ecosystem with endpoints is completly unattractive for any Softswitch Manufacture (Cisco, Siemens, Avaya, etc.) Why they should help the biggest and toughest competitor.

    Snom has faced all of this challenges and a lot more... If you dont like the 840$ Tanjay, try to go with snom OCS Edition (also compatible to Siemens Hipath 8000! or any other SIP System). It will offer you a broad range of phones (5 + the new MeetingPoint Conference Solution). If you like, you can failover for Snom from OCS to Siemens (Cisco, Nortel, Asterisk etc.) via one or more fallback Identity in case of any outage at OCS.

    Best regards,



    Jan Boguslawski | Consultant IT Infrastructure | MCSE, MCTS OCS | ITaCS Berlin | www.itacs.de
    Saturday, February 21, 2009 7:24 AM
  • Griffter,

    Next month we will be releasing a product that allows you to connect any standard SIP phone to OCS as well as use any ITSP of your choice with OCS rather than the small number that are supported.  I'll post on my blog when it is released (link below) and will also try to post an update in the forums as well.

    Mike Stacy | Evangelyze Communications | http://www.evangelyze.net/cs/blogs/mike
    Saturday, February 21, 2009 3:49 PM
  • Just one comment: OCS and OC 2007 (RTM or R2 release, either one) can and *do* support a wide variety of codecs and bit rates, including (at least, and going from memory here) G.711 a and mu law, Siren, G.722.1, G.723.1, G.726 and GSM.

    Most existing SIP phones can't natively register with OCS due to limitations in their security and authentication functionality. RTAudio support is NOT a blocking issue, though you'll hear that from lots of people.

    Saturday, February 28, 2009 12:38 AM
  • Thanks Mike - look forward to the blog!
    Tuesday, March 3, 2009 9:13 PM
  • Let's be clear although the PBX world moved to SIP for endpoints / trunks and other gateways, Microsoft has chosen their own way to enter the VoIP arena. Adding a codec that nobody uses, RT Audio, and the need for a mediation server in a 1:1 fashion for every voice gateway, are some examples. No word about if this is wrong or bad but for sure it is not the route to interoperability and open standards.
    To be open and give customers a choice it would have been much better that OCS would support SIP endpoints according to the known RFC's. With this any SIP endpoint could have been connected, the "proprietary" codec and some other reasons makes this a no go. Microsoft supports G.711/729 but that's on SIP trunks and not endpoints, they use the mediation server to convert their RT Audio into the open standards based codecs the PBX/VoIP world is using.

    Maybe this will change in future......
    Sunday, March 8, 2009 3:28 PM