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Cannot find Server URI or Line URI for EV w/PBX Integration RRS feed

  • Question

  • Hi guys. I'll preface myself by saying I'm basically ignorant of phone systems & OCS. This project was dropped in my lap, and in between the 133 page VoIP guide, the 100 page Deploying Enterprise Voice guide and the 100 page Deploying Standard Edition guide (not to mention all the internet sites I've scoured), I'm just feeling way in over my head.

    I cannot, for the life of me, get the phone integration to work. The Telephony guys here aren't a lot of help since they also have no experience with this. Asking them for the Server URI of the phone system basically gets blank stares and an IP address in response.

    Can anyone help me in any way? Where can I find this information for our environment? It has to be somewhere, right?

    Wednesday, August 12, 2009 6:01 PM

Answers

  • Hello
    I'm sorry.. I mistook that sip:username@rcc_gateway_address is for Server URI not Line URI.

    Line URI is used only OCS. This is Communicator user's extention number.
    OCS only supports E.164 format at SIP URI (RFC 3966).
    For example tel:+1-444-555-6666. You can also add extention number as tel:+1-444-555-6666;ext=6666 and so on.
    But, in many case you don't need to configure extention number.
    In my experience, SIP URL shoud type E164 Global address format (ex. :+1-444-555-6666) not tel:6666.
    If you type tel:6666, CallerID will shown as "Unknown Number"

    Server URI is for login account at RCC gateway. RCC gateway meens  translate CSTA/SIP protocol to CTI/QBE protocol.

    RCC connection is as follow.


    Microsoft's Phone Integration is 3 type.

    1.Communicate to PBX or PSTN
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---Basic Gateway---PSTN
    or
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---PBX

    2.RCC(Remote Call Controll)
    RCC can control PBX Phone (Mitel Phone) from Office Communicator remotely, 
    Communicator---OCS---<CSTA/SIP>---RCCgateway---<CTI/QBE>---PBX---Phone

    3.Dual forking
    Dual forking meens Ring the PBX phone and Communicator at same time.
    * PBX must support forking controll to SIP network

    Mitel guys told me, Mitel 3300 mxe does NOT support RCC nativly,But can support with Mitel Live Business Gateway

    Please refer as following
    http://www.mitel.com/DocController?documentId=9506
    http://www.mitel.com/DocController?documentId=15000

    Good Luck


    YamamotoA


    Bah!!! Why does the Microsoft documentation show the 3300 supporting a direct connection to OCS 2007 if doesn't natively support RCC? This is why I hate doing these projects. It's impossible to find this information. If I knew I needed LBG for OCS 2007, I never would have upgraded from LCS 2005. That was the whole POINT of upgrading - to get away from LBG (which is a terrible, terrible piece of software, BTW. We spent MONTHS working with Mitel to try to get it working, and never got any functionality out of it besides toast on incoming calls - Mitel charged us for the software & man-hours to get it running, and eventually gave up saying, "We can't figure it out. Something in your environment is interfering with LBG. Sorry. But thanks for the fat check."

    If it's true that we need LBG for RCC, I'll probably just dump this project and tell my boss it's not possible.

    Then again, I'm so confused about what provides what functionality, it might not matter at all. It sounds like I'll get all the functionality we need from an Enterprise Voice with PBX integration – blank SIP Server URI (aka Dual Forking).

    My confusion is only this: In this set-up, when a person gets a call, they get the "toast" on Communicator. If they pick up their IP phone, Communicator DOESN'T change their state to "in a call." That is what it is... But if someone clicks the "Answer" button in Communicator - it DOES change their state to "in a call." But when you click the Answer button - where does the voice stream get routed to? The PC? The IP Phone? It doesn't say.

    ALSO... It says in that set up, a Media Gateway is required - but NOT a Mediation Server... Is that correct?


    Enterprise Voice with PBX integration, I wonder if SIP URI is configured to non E.164 format like tel:6666 and Mitel phone extention is 6666. right?
    This case, you may be failed to Dual forking.
    I think reconfigure SIP URL to tel:+1-444-555-6666 and Mitel Phone 6666, and configure Mitel converts forking target  to +1-444-555-6666.
    Please ask Mitel Guys chat.

    Media Gatey is required. Because Media Gateway translate SIP signaling and Audio codec to PSTN signaling.
    But If Mitel 3300 has PSTN ports like T1 or E1, You don't need Media Gateway. Mitel 3300 is just Media Gateway.

    And Mediation Server, Does Mitel 3300 supports SIP/TLS and RTAudio? If it supports these protocol, no Mediation Server need.

    Do you understand me?

    YamamotoA
    Wednesday, August 19, 2009 4:40 PM

All replies

  • The Server URI field (msRCTSIP-OptionFlags) is basically only used in Remote Call Control scenarios.  For standard Enterprise Voice functionality you'll just need the Line URI (msRTCSIP-Line) populated with a user's number, typically a DID. Extensions can be used if unique DIDs are not available for each individual user.  Example values would be "tel:+13125551212" or "tel:+13125551200;ext=1212"

    Take a look at this excellent blog article which highlights the different Voice configurations:
    http://blogs.technet.com/jkunert/archive/2008/07/30/voice-scenarios-with-ocs-2007.aspx

    And then I have some deeper details on the those specific AD attributes here:
    http://blogs.pointbridge.com/Blogs/schertz_jeff/Pages/Post.aspx?_ID=20

    Jeff Schertz, PointBridge | MVP | MCITP: Enterprise Messaging | MCTS: OCS
    Wednesday, August 12, 2009 11:02 PM
    Moderator
  • Unfortunately, I'd stumbled across those fields just poking around in ADSIedit, and it seems those fields just reflect what is already in the OCS user properties. If I change the fields for Server URI or Line URI within OCS, it changes those fields.

    So, how do I know what should be there to begin with??

    For users who have never had voice enabled, it simply shows <not set> in ADSIedit. There has to be a place I can look up what those fields SHOULD BE.
    Thursday, August 13, 2009 1:01 PM
  • To be specific, when I launch Communicator, in the Notifications area I get a message that says "No Phone System Connection." Further details: "Cannot connect to the phone system. The call control server may be temporarily unavailable. If the problem persists, please contact your system administrator."

    Thursday, August 13, 2009 6:20 PM
  • Hi

    Per your description, you are planning to deploy the OCS included Enterprise Voice. Sure, it is a big project; you must deploy the OCS SE, the Mediation Role server, the VOIP gateway and configure many settings and so on. Yet, if you want to deploy the OCS SYSTEM by yourself and your PBX partner, I think youd better do some research about the OCS enterprise voice firstly.

    For your first question, Jeff gave a good answer in the blog. You can get a detail instruction for it.

    Here are some other instructions about the Enterprise Voice you can refer to.

    About the planning for Enterprise Voice you can refer to below links:

     http://technet.microsoft.com/en-us/library/dd441292(office.13).aspx

    http://technet.microsoft.com/en-us/magazine/2008.02.ocstelephony.aspx

    About how to deploying Enterprise Voice you can refer to below links:

    http://technet.microsoft.com/en-us/library/dd441382(office.13).aspx

     

    For your last question, have you deployed the Enterprise Voice in your system? If you did, could you please publish some event log of the OC? So we can do more research regarding your issue.

     

    Hope this helpful!

    Regards!

    Wednesday, August 19, 2009 6:29 AM
    Moderator
  • Hello.
    In order to configure Phone system integration (RCC) , your PBX MUST support CSTA/SIP protocol.
    In Cisco case, Callmanager is phone system, but it doesn't support CSTA/SIP its self. But with Cisco Unified Presence connects OCS to Callmanager.

    RCC configration summary is as follow.

    On OCS
    1. User's Line URI sip:username@RCC_gateway_adrress
    2. Route to RCC gateway
    3. Autorize RCC gateway adrress

    On RCC gateway (if need)
    1. Accept to CSTA/SIP protocol from OCS
    2. Configure RCC user account
    3. Redirect to Phone System

    On Phone System
    1. Map user and Phone number

    Thanks for your patience my broken English 


    YamamotoA
    Wednesday, August 19, 2009 8:02 AM
  • You guys will forgive me for my ignorance, but continuing to get conflicting (seemingly) answers is really not what I'd hoped for.

    I just want to know what goes in the Line URI field, and what goes in the Server URI field - where I can find that information. The first article Jeff linked to show someone using tel:+1(tendigitnumber) or tel:+1(tendigitnumber);ext=(fourdigitextension) - and then the next article shows tel:(4digitextension);phone-context=dialstring... Where does THAT come from, and which one is correct??

    Then Akira says to use sip:username@rcc_gateway_address for the Line URI (which I think he's just mistaken for the Server URI, in which case it could be helpful, but I don't know what the RCC gateway is. Is that my mediation server?? Or the PBX?)...

    I've read ALL of those documents, but not having a background in telephony OR unified communications OR Exchange, a lot of it just makes me dizzy, frankly.

    We have a Mitel 3300 mxe, which is shown as directly supporting SIP:
    http://technet.microsoft.com/en-us/office/bb735838.aspx

    I've deployed a Mediation Server, but not a media gateway (as the documentation was VERY confusing and it seemed like we wouldn't need one since the PBX directly supports SIP).
    Wednesday, August 19, 2009 2:42 PM
  • The Line URI should be "tel:" followed by the DID number assigned to the Enterprise Voice user.  Typically that value is something like "tel:+13125551234" but it depends on your telephony configuration.

    The Server URI is only used for Remote Call Control and points to a CTSA server, as in "sip:username@nexthop.domain.com" where the RCC next hop and remote user account name are used. 

    And since the Mitel 3300 is certified for Direct SIP, then yes you need a Mediation Server only, no separate media gateway.


    Jeff Schertz, PointBridge | MVP | MCITP: Enterprise Messaging | MCTS: OCS
    Wednesday, August 19, 2009 3:43 PM
    Moderator
  • Hello
    I'm sorry.. I mistook that sip:username@rcc_gateway_address is for Server URI not Line URI.

    Line URI is used only OCS. This is Communicator user's extention number.
    OCS only supports E.164 format at SIP URI (RFC 3966).
    For example tel:+1-444-555-6666. You can also add extention number as tel:+1-444-555-6666;ext=6666 and so on.
    But, in many case you don't need to configure extention number.
    In my experience, SIP URL shoud type E164 Global address format (ex. :+1-444-555-6666) not tel:6666.
    If you type tel:6666, CallerID will shown as "Unknown Number"

    Server URI is for login account at RCC gateway. RCC gateway means  translate CSTA/SIP protocol to CTI/QBE protocol.
    CSTA/SIP and CTI/QBE is Phone Remote Control message.

    Microsoft's Phone Integration is 3 type.

    1.Communicate to PBX or PSTN
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---Basic Gateway---PSTN
    or
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---PBX

    2.RCC(Remote Call Controll)
    RCC can control PBX Phone (Mitel Phone) from Office Communicator remotely,
    Communicator---OCS---<CSTA/SIP>---RCC gateway---<CTI/QBE>---PBX---Phone

    3.Dual forking
    Dual forking means Ring the PBX phone and Communicator at same time.
    * PBX must support forking controll to SIP network.

    Mitel guys told me, Mitel 3300 mxe does NOT support RCC nativly,But can support with Mitel Live Business Gateway.
    Mitel Live Business Gateway acts as RCC gateway


    Please refer as following
    http://www.mitel.com/DocController?documentId=9506
    http://www.mitel.com/DocController?documentId=15000

    Good Luck

    Wednesday, August 19, 2009 3:58 PM
  • Hello
    I'm sorry.. I mistook that sip:username@rcc_gateway_address is for Server URI not Line URI.

    Line URI is used only OCS. This is Communicator user's extention number.
    OCS only supports E.164 format at SIP URI (RFC 3966).
    For example tel:+1-444-555-6666. You can also add extention number as tel:+1-444-555-6666;ext=6666 and so on.
    But, in many case you don't need to configure extention number.
    In my experience, SIP URL shoud type E164 Global address format (ex. :+1-444-555-6666) not tel:6666.
    If you type tel:6666, CallerID will shown as "Unknown Number"

    Server URI is for login account at RCC gateway. RCC gateway meens  translate CSTA/SIP protocol to CTI/QBE protocol.

    RCC connection is as follow.


    Microsoft's Phone Integration is 3 type.

    1.Communicate to PBX or PSTN
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---Basic Gateway---PSTN
    or
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---PBX

    2.RCC(Remote Call Controll)
    RCC can control PBX Phone (Mitel Phone) from Office Communicator remotely, 
    Communicator---OCS---<CSTA/SIP>---RCCgateway---<CTI/QBE>---PBX---Phone

    3.Dual forking
    Dual forking meens Ring the PBX phone and Communicator at same time.
    * PBX must support forking controll to SIP network

    Mitel guys told me, Mitel 3300 mxe does NOT support RCC nativly,But can support with Mitel Live Business Gateway

    Please refer as following
    http://www.mitel.com/DocController?documentId=9506
    http://www.mitel.com/DocController?documentId=15000

    Good Luck


    YamamotoA


    Bah!!! Why does the Microsoft documentation show the 3300 supporting a direct connection to OCS 2007 if doesn't natively support RCC? This is why I hate doing these projects. It's impossible to find this information. If I knew I needed LBG for OCS 2007, I never would have upgraded from LCS 2005. That was the whole POINT of upgrading - to get away from LBG (which is a terrible, terrible piece of software, BTW. We spent MONTHS working with Mitel to try to get it working, and never got any functionality out of it besides toast on incoming calls - Mitel charged us for the software & man-hours to get it running, and eventually gave up saying, "We can't figure it out. Something in your environment is interfering with LBG. Sorry. But thanks for the fat check."

    If it's true that we need LBG for RCC, I'll probably just dump this project and tell my boss it's not possible.

    Then again, I'm so confused about what provides what functionality, it might not matter at all. It sounds like I'll get all the functionality we need from an Enterprise Voice with PBX integration – blank SIP Server URI (aka Dual Forking).

    My confusion is only this: In this set-up, when a person gets a call, they get the "toast" on Communicator. If they pick up their IP phone, Communicator DOESN'T change their state to "in a call." That is what it is... But if someone clicks the "Answer" button in Communicator - it DOES change their state to "in a call." But when you click the Answer button - where does the voice stream get routed to? The PC? The IP Phone? It doesn't say.

    ALSO... It says in that set up, a Media Gateway is required - but NOT a Mediation Server... Is that correct?

    Wednesday, August 19, 2009 4:15 PM
  • Hello
    I'm sorry.. I mistook that sip:username@rcc_gateway_address is for Server URI not Line URI.

    Line URI is used only OCS. This is Communicator user's extention number.
    OCS only supports E.164 format at SIP URI (RFC 3966).
    For example tel:+1-444-555-6666. You can also add extention number as tel:+1-444-555-6666;ext=6666 and so on.
    But, in many case you don't need to configure extention number.
    In my experience, SIP URL shoud type E164 Global address format (ex. :+1-444-555-6666) not tel:6666.
    If you type tel:6666, CallerID will shown as "Unknown Number"

    Server URI is for login account at RCC gateway. RCC gateway meens  translate CSTA/SIP protocol to CTI/QBE protocol.

    RCC connection is as follow.


    Microsoft's Phone Integration is 3 type.

    1.Communicate to PBX or PSTN
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---Basic Gateway---PSTN
    or
    OCS---<SIP/TLS,RTAudio>---Mediation Server---<SIP/TCP, G.711 etc.>---PBX

    2.RCC(Remote Call Controll)
    RCC can control PBX Phone (Mitel Phone) from Office Communicator remotely, 
    Communicator---OCS---<CSTA/SIP>---RCCgateway---<CTI/QBE>---PBX---Phone

    3.Dual forking
    Dual forking meens Ring the PBX phone and Communicator at same time.
    * PBX must support forking controll to SIP network

    Mitel guys told me, Mitel 3300 mxe does NOT support RCC nativly,But can support with Mitel Live Business Gateway

    Please refer as following
    http://www.mitel.com/DocController?documentId=9506
    http://www.mitel.com/DocController?documentId=15000

    Good Luck


    YamamotoA


    Bah!!! Why does the Microsoft documentation show the 3300 supporting a direct connection to OCS 2007 if doesn't natively support RCC? This is why I hate doing these projects. It's impossible to find this information. If I knew I needed LBG for OCS 2007, I never would have upgraded from LCS 2005. That was the whole POINT of upgrading - to get away from LBG (which is a terrible, terrible piece of software, BTW. We spent MONTHS working with Mitel to try to get it working, and never got any functionality out of it besides toast on incoming calls - Mitel charged us for the software & man-hours to get it running, and eventually gave up saying, "We can't figure it out. Something in your environment is interfering with LBG. Sorry. But thanks for the fat check."

    If it's true that we need LBG for RCC, I'll probably just dump this project and tell my boss it's not possible.

    Then again, I'm so confused about what provides what functionality, it might not matter at all. It sounds like I'll get all the functionality we need from an Enterprise Voice with PBX integration – blank SIP Server URI (aka Dual Forking).

    My confusion is only this: In this set-up, when a person gets a call, they get the "toast" on Communicator. If they pick up their IP phone, Communicator DOESN'T change their state to "in a call." That is what it is... But if someone clicks the "Answer" button in Communicator - it DOES change their state to "in a call." But when you click the Answer button - where does the voice stream get routed to? The PC? The IP Phone? It doesn't say.

    ALSO... It says in that set up, a Media Gateway is required - but NOT a Mediation Server... Is that correct?


    Enterprise Voice with PBX integration, I wonder if SIP URI is configured to non E.164 format like tel:6666 and Mitel phone extention is 6666. right?
    This case, you may be failed to Dual forking.
    I think reconfigure SIP URL to tel:+1-444-555-6666 and Mitel Phone 6666, and configure Mitel converts forking target  to +1-444-555-6666.
    Please ask Mitel Guys chat.

    Media Gatey is required. Because Media Gateway translate SIP signaling and Audio codec to PSTN signaling.
    But If Mitel 3300 has PSTN ports like T1 or E1, You don't need Media Gateway. Mitel 3300 is just Media Gateway.

    And Mediation Server, Does Mitel 3300 supports SIP/TLS and RTAudio? If it supports these protocol, no Mediation Server need.

    Do you understand me?

    YamamotoA
    Wednesday, August 19, 2009 4:40 PM
  • *sigh*

    I just spoke to the telephone guys, and apparently we need another appliance to do the dual forking. At this point, I'm wondering exactly what part of the 3300 is compatible with OCS, because if I need another server for RCC, and an appliance for the forking, it doesn't sound like its natively compatible IN ANY WAY!

    I think that just was the final wrench in the works here. Thanks for your help guys. I'll post back if my boss decides he wants to spend MORE money on an appliance or server...

    Wednesday, August 19, 2009 6:11 PM