I have a BCM400 which suddenly crapped out after years of service. I pulled the BFT and it doesn't seem to be part of the group that had the capacitor issue. I hooked up a monitor and powered it up, and here's what I see on the display once the boot sequence runs its course:
OS Loader V4.01...
I'm not familiar with CME, but on CUCM there is a drop-down on the 7970 configuration page for an expansion module. If it is not set for the 7914, you get the Christmas-tree result that you're seeing. I would poke around the 7970 configuration on CME to see if there is something for an...
I don't know what platform you're on, but on CUCM for example, there is a Voice Mail configuration section that has two or more MWI DNs - one for on and one for off. You can turn it off by simply dialing the appropriate DN from the phone. Note that you may need to modify your calling search...
Here's one approach:
1. Set up two partitions (let's call them INBOUND-EXT-PT and INBOUND-INT-PT).
2. Put the DN 3114 on the phone in the INBOUND-INT-PT partition.
3. Set up a translation on DN 3114, but in the INBOUND-EXT-PT partition, and have it point to 6953.
4. Set up a calling search space...
Perhaps you need an MTP? You can find this and other hints in the SRND for CUCM 8:
Media termination points (MTPs) are generally not required for normal operation of the H.323 trunk.
They are, however, required for communication with devices that are H.323 Version 1, that do not support the...
I can't think of anything specific, other than keeping your fingers crossed as you'll find that no two vendors' implementations of SIP are alike. We've even found instances where upgrading an application to a different release from the same vendor can present challenges on how SIP calls behave...
I am trying to initiate the Live Record feature with a simple press of a button. However, I'm having some challenges. The idea is to have an xml or jsp application that will perform four keystrokes - Conference, #, #, Conference. Using examples that I have come across, I was able to set up a...
...or possibly you just don't have enough inbound traffic that this particular PRI gets hit with linear hunting. You could always test by busying out the ones in front of it in the hunt sequence and verify that calls are coming in, if you're worried that it may not be working.
Older CMs don't support SIP directly. You'll need to set up a Session Border Controller or CUBE between the two, and run h323 or an Inter-Cluster Trunk to the SBC/CUBE, then SIP from there to your other PBX.
It's a little counter-intuitive, but you'll need to scroll down to where you see a DHCP = Yes prompt and change it to No. That should unlock the address field at the top of the section so that you can scroll back up there and change it to whatever you want.
One thing I just thought of - basically the system is initiating a transfer, which will initially send the DN of the person (or in this case the CallPilot port) that initiates the transfer, then the transfer is completed. When a call goes out a PRI trunk, there is an initial d-channel message...
I have Vonage working via an FXO port on a Nortel BCM, and it does provide caller ID. If your configuration supports caller ID from the PSTN, then it should support it from Vonage. I wouldn't think that this is a Vonage-specific issue for Cisco or any other equipment.
I have seen occasional issues where users don't have admin rights on the local machine. We have resolved this by either granting the user local admin rights, or having an admin log onto the machine, running Web Client, then logging back off. After that things seem to work right again.
I have not been able to duplicate the issue myself, and I haven't found that max calls had any effect (it is set high for these phones anyway because they're too lazy to use our conference bridge service).
I opened a TAC case and got a recommendation to either turn off Privacy or use the Barge...
Thanks for your input. I suspect that this is indeed user error, but I have heard this from the admins of two high-ranking executives, and I need to make sure that I have looked into this thoroughly.
Here's the scenario -- a number of our users have shared DNs, where admins will place a call or set up a conference call on a shared line at an executive's request, then place it on hold so that the executive can pick it up on his/her phone. They inform us that on occasion, they are not able to...
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