locked
Remote Call Control RRS feed

  • Question

  • Hello,

     

    We are trying to get the remote call control to work with a sip PBX and are a little confused on how the call is supposed to flow. We cuurently have a mediation server in place and can place a PSTN call though the mediation server. Does the remote call control meesages flow the same way? Are there other pieces that need to be in the mix for us to get the remote call control to work?

     

    Also if there are any sugestions on how the sip server uri and the line uri should be formated it would be much appreciated.

     

    Thanks,

    Mike

    Tuesday, July 17, 2007 2:19 PM

Answers

  • The next hop is still there, you have to enter the cups IP and the domain you want to put in the users' RCC server URI parameter, and you have to authorize the host (cups IP too). In the user tel uri, if I remember well (I don't config with me) I put something like tel:1000;phone-context=dialstring (found in the cups doc ver 6).

    the most config is on the cups, but I just follow the cisco doc, restart CCM and CUPS services,  and that's all.

     

     

    Friday, July 27, 2007 7:19 AM

All replies

  • Call controll message from OC to Mediation use CSTA-SIP protocol. Now I want understand which is the protocol that you use for understanding call cantroll messagge? Is your PBX able to manage CSTA-SIP message?

     

    Wednesday, July 18, 2007 10:06 AM
  • I am looking into that now. Our Polycom phones are using Interactive Intelligences (www.inin.com) sip proxy as the registration server. I assume that needs to be able to handle the CSTA messages, correct?
    Wednesday, July 18, 2007 10:13 PM
  • I don't know your solution.

    If you want that Call Controll works:

    1. Your PBX must support Call Controll;

    2. Your Call Controll must support CSTA message;

    About second point if your application don't supports CSTA protocoll but supports another protocoll then you must install a third party application that traslate the CSTA protocoll in your protocoll. Ex. OCS support CSTA and your solution TAPI --> you have need a Controll Gatway that translate TAPI message in CSTA message.

     

     

    Thursday, July 19, 2007 8:07 AM
  • Hi

     

    I did a few RCC integrations, with Alcatel GETS (Genesys Entreprise Telephony Service) and also with CUPS (Cisco Unified Presence server). These third party applications allow OCS to retreive telephony events from your PBX, and send them as SIP messages to Communicator, for example when the phone receives a call, to toaster to pop up in Communicator, with name and phone nb from caller.

    Your PBX must support any phone control protocol, CSTA is the most common one, start this check first. For SIP pbx like Asterisk, I've heard some projects for phone control protocol, but never investigate.

    Sunday, July 22, 2007 11:52 AM
  • @nkv: do you know some documentation for rcc?! i know how to do it on lcs, but for ocs i didn't find anything...
    Wednesday, July 25, 2007 1:25 PM
  • Hi,

     

    RCC is almost the same configuration between LCS and OCS. On which PBX or CTI middleware are you working on ?

    For CUPS I found documentation on Cisco web site. For GETS it's more tricky.

     

    Thursday, July 26, 2007 3:02 PM
  • on lcs there was a tel next hop configuration tab - thats what i miss on ocs

    i use ccm5.1/cups1.3.
    cups configuration is clear. but the ocs side i'm still working on...
    Thursday, July 26, 2007 3:05 PM
  • The next hop is still there, you have to enter the cups IP and the domain you want to put in the users' RCC server URI parameter, and you have to authorize the host (cups IP too). In the user tel uri, if I remember well (I don't config with me) I put something like tel:1000;phone-context=dialstring (found in the cups doc ver 6).

    the most config is on the cups, but I just follow the cisco doc, restart CCM and CUPS services,  and that's all.

     

     

    Friday, July 27, 2007 7:19 AM
  • You mean the Next Hop Connections Tab in Mediation Server Properties?!

    There you can configure only one PSTN next hop. So when i want to use both Enterprise Voice (SIP Trunk to Callmanager) and RCC (SIP/CSTA Trunk to CUPS) i have to decide what feature is more important to me...

    Can you confirm this?!
    Friday, July 27, 2007 8:55 AM
  •  

    Hi

     

    You don't need Mediation server for RCC, all is in the pool and front end server properties: routing and host authorization, the same as LCS.

     

    Monday, July 30, 2007 7:32 AM