Search results for query: *

Have you hard coded the CM address in the phone or set option 150 for the DHCP options at the remote site? Here is the step by step troubleshooting guide. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8941_8945/8_5/english/admin_guide/8941_8945adm851/6921trb.html

MTP - Media Termination Point, it's needed for SIP and H323 gateways.

Is this VG configured as the MTP for the device pool? I saw those errors, but will commonly see some after the router reboots and calls are attempting to come in while the router is trying to register. Is this a copper PRI, or is this a handoff to a local device?

If the show isdn status output looks the same as it does now, as in Multiple Frames Established, then the issue isn't with the PRI itself. 1. Would you clarify what you mean when you say their "phones drop"? Are they dropping the current caller, does the phone reset? 2. Is this a VG to CUCM or...

Rebooting the router is quite extreme to resolve an issue with a PRI. Have you tried just shutting down the interface itself and then turning it back on? Additionally what is the output of "show isdn status" when the Pri stops working?

This is a prime example of why its recommended that you protect a voice gateway with a dedicated firewall or at the very least a Zone Based Firewall. Usually calls such as this are generated by script kiddies and serve no real purpose other than to run up your bill. Additionally they are also...

In your case your only option would be Static Nat. However, the bigger question is why isn't your network segmented at the colo? Each client of a colo should be in a separate VRF in which there would be no IP overlap and no nat necessary.

Check your licenses on the module.

1. Why are you still on Call Manager version 4? 2. Doesn't the Business Name already display via callerid?

Do what? Sip is a gateway issue not a UCCX issue.

The fact you refuse to adapt to a clients request indicates you are going out of business very soon. You are punching to a 66 block because its all you know and you fail to gain new skills. Every single issue you have tried to raise is because of a bad install and not a bad product. I have no...

Why not get an ethernet based connection and do SIP natively?

You would be better off with a VG202 or VG224.

I should have added the best way to test this is to try "show dialplan number 2222" and it will show you where the call is going to route. Additionally look at the voice translation-rules that should be at the top of the configuration.

Burt, I tried to follow your post, but got rather confused. You say they are transferring a call to 2222, but you show a debug isdn q931 that shows the call hitting the pri? Most likely the voice mail box number matches part of the dial string they are trying to transfer towards. For example...

You have to setup a transcoder and specify the connection on the dial peer.

There isn't a way other than typing them all out.

Ditch the router and get an ASA 5505 for a few hundred dollars.

You would be better off with a DMVPN or GETVPN.

You have to download the new IOS from Cisco with a valid account.