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I have used TELSTRAT's solutions for call recording on TDM and IP. I personally like the way they integrate to the CS1000. I guess is worth for you to take a look in their webpage.

First of all take a look to all the features that the M2250 offers and see how many of them you are using right now. This analysis will tell you if you need a console or a telephone can fulfill your requirements. In my experience most of the times the operators are only transfering calls wich...

Your MGC should be cabled: 1E to ELAN Switch 2T to TLAN Switch CE and CT are for direct conection to CCPM CS and SS respectively.

The following NTPs will help you understand where you are moving to: - IP Peer Networking Installation and Commissioning - Network Routing Service Installation and Commissioning - Converging the Data Network with VoIP Fundamentals However if you are creating a Network I would seek advise from...

CSSMT, 10.0.32.12 is your SS but at the same time is your NRS because you have both applications (SIP Gateway and NRS) in the same server A normal call should have the following (summarized) process: 1. SIP GW sends an INVITE message to NRS (If NRS and SIP GW are in the same serverm you will...

Remember that the node information is stored in the Call Server, therefore if you want to start from scratch first you must delete the node from Element Manager if not every time you reboot the CS will re display the stored information

Goutam, Did you assign the new zone (VTRK) to route 1? Is Signaling Gateway configured as TCP for SIP? When you dial 601008 at what point you get fast busy? Before or after you finish dialing all digits? You never provided a printout for the SIPCallTrace I recommended. Can you post the results?

You have route 1 in Zone 000, normally zone 000 is the default zone and is configured with zone intent = MO. Check zone configuration parameters in LD 117 or via Element Manager. Create a new zone with zone intent = VTRK for your route 1 (SIP Route)

Your ITG IP Trunk licenses will be converted to H323 or SIP trunks when you engineer the upgrade. The MC32 Cards will still provide DSP resources

If you say that the call is reaching the SIPEX that means that the NRS is doing its job and sending the call to the correct endpoint. Check codecs and payload you have enabled in the SS and the SIPEX.

CSSMT, I suggest you enable a SIPCallTrace via CLI in the Signaling Server and share what messages you receive during the call attempt from the CS1000 to the CM. SIPCallTrace on traceAllOn If you don't see any messages then your CS1000 is blocking the call before sending it to the SIP Route in...

Try reformatting the CPPM Call Server FMD (1GB Compact Flash) and reload SW. If that doesn't work then replace the 1GB Card.

Just a note, remember to INI after you change IP Addresses. If not and you STAT ROUT and STAT HOST you will still see your old data around.

Assuming the ISP1100 is the Primary NRS and there is no Alternate in your network I would do this: 1.- Change NRS Role from Primary to Alternate 2.- Change the information at each SS level so: -Primary NRS IP Points to a dummy IP -Alternate NRS IP points to ISP1100 * These 2 steps are service...

Crac74, Take a look to the SIP URI map in the Signaling Server and make sure that it matches the info in the NRS L0/L1 Domains. SS Private/UDP domain name = udp Private/CDP domain name = cdp.udp NRS L0 Domain = cdp L1 Domain = udp According to the DCH messages the call is properly marked as...

Carb11: What about the IP Addresses of the 2050, the 1140 and the Signaling Server TLAN? Are the 3 elements in the same subnet or different subnets? If they are in different subnets, check all default gateways for the 3 elements including Signaling Server.

If it is a SIP Trunk you can do a SIPCallTrace in the Signalling Server and you will have some more info on that call.

When do you get reorder tone? Once you finish dialing or before you enter the last digit? Do you get reorder when dialing to another IP set or when dialing a TMD set or TDM trunk? Can you receive calls on the 2050?

Mike, Yes 83xxx is a properly configured routing entry in the NRS. In fact if I do the SIP routing test to 83xxx I get the correct gateway endpoint.

Good day all, I hope you can give me some light on this problem. The call scenario is as follows: Node A: CS1000E, CCPM, R500W + SS CCPM R5.00.31 SS TLAN IP: 172.25.15.20 Node IP: 172.25.15.50 SS Apps: SIP GW, H323GW, NRS (Primary) Calling DN: 84177 Node B: Static SIP Endpoint IP...