I'm hoping you can help point me in the direction of the correct documents to help me complete the integration of the above.
Currently we have a successfully running OCS 2007 server for PC to PC voice & video (externally too with the OCS Edge Server), however I am slightly lost and confused when it comes to full integration with our Call Manager and CUPS Server.
Please bear with me on this as I am a network engineer and have had only very little exposure to Call Manager and Telephony (our Telecoms Dept completed the installation of CUCM 6 & Cisco 7970's)..
Below I have listed the functionality i am looking for, perhaps you can confirm what is indeed possible:
1 - Outbound calls from IP Phone (already working) and OC Client via ISDN30
2 - Inbound DDI calls via ISDN30 routed to OC Client
3 - Busy/On Call indicator when on either an OC call or Cisco IP Phone is being taken
4 - Ability for both OC Client and Cisco IP phone to ring when a call is made (is that dual-forking?)
5 - Will external calls via OC Client work when a user is "off network" ie. at Home?? (currently this works with PC to PC calls as it routes through Edge Server, could phone calls use this route?)
6 - Call re-direction in OC Client
Our Cisco Router is a 2851...
I think in essence i am looking for Enterprise Voice, not just RCC...
I'm confused as to where the Mediation Server comes in and when I need to use the CUPS server in this configuration.
1. Outbound calls from OCS need to head to the Cisco 2851 via a SIP Trunk. The 2851 needs to have a dial-peer setup with the pattern to strip the + then send it out the PRI or PSTN connectivity. Or Strip the + and send it to CUCM where it process the call. (not sure if you are MGCP or H323 on the 2851 gateway.
2. Inbound calls work similar. You need route the traffic through the SIP trunks. In CUCM 6.x, its a little tough and you need the 2851 to do this. It does work directly, but some features seem to be missing if Im not mistaken. CUCM 7, you can use it directly with OCS. (have not tested myself)
3. Presence of the call is only controlled by OCS/CUPS server. So you need to create those links for it to work. Basically, CUCM monitors all lines (of course) and when the SUBSCRIBE CSS is configured on the line, it will send out on/off hook status to the CSS. The SIP connection to presence will relay this to OCS server and onto the OCS Client.
4. Dual forking is done by Mobility now. It's not really dual-forking, but using the feature of "Mobility" and in CUCM 7.0 it works much better. (have not tested yet)
5. As long as you are routing the IPs correctly, it should work in theory. It's just a SIP connection. But not knowing the rest of the setup, proxy, NAT translations, etc.. it would need to be mapped out (call flow included)
6. Call re-direction. Do you mean placing the OCS ringing client on fwd?
SIP traces provide key information in troubleshooting SIP Trunks, SIP
endpoints and other SIP related issues. Even though these traces are in
clear text, these texts can be gibberish unless you understand fully
what they mean. This document attempts to br...
Please find the attached HTML document, download and open it on your PC.
This provides an easy to use form where you simply answer a few
questions and it will render the proper jabber-config.xml file for you
to copy/paste. There is built in logic to verif...
CUCM Database Replication is an area in which Cisco customers and
partners have asked for more in-depth training in being able to properly
assess a replication problem and potentially resolve an issue without
involving TAC. This document discusses the bas...