I need help and your suggestion in configuration SIP trunk configuration on C3945-CME-SRST/K9 router. We are running CUCM version 8.0.3 and we currently have two PRI's on our voice gateways to communicate over to PSTN but we want to add one more SIP trunk with 6MB bandwidth available to us for an additional DID range. I need to know what should be the configuration required on Call manager side and what configuration would be requried on C3945-CME router. If you direct me to any link where information is available and also what should be my starting point on this project. My telco have provide me following configuration to be used
Circuit is activated in SS and number range is ready for customer use. xxxxxxx100-xxxxx999
On CUCM you would build a SIP trunk to the gateway (not the ITSP) and add it to your route group(s) if you want it to process outgoing calls. Be sure to ask your providers whether they filter ANI on an outbound call. In other words, if they will only accept outbound calls from their own DIDs. If they do, you'll have extra work on your hands to split the phones into separate CSS with their own route patterns, etc. Almost like a separate site in CUCM.
On the gateway you need to purchase CUBE if you haven't already. It's purchased per concurrent call you want to support, such as 25 calls. The final step is to get dial-peers, voice service voip, and sip-ua appropriately configured. This is probably the steepest part of the learning curve.
Here is some CUBE reference material to get you started:
These are the paths to get to each CCX logs through CLI. They may be helpful if you are having issues accessing RTMT or downloading logs through it.
If you want to download them you have to prefix "file get " and you can add one of the options (re...