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- About jmunoz19
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Member since
01-25-2009
146
Posts
54
Helpful
20
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01-15-2016
12:45 PM
Hi guys,
We have a N7K VPC solution in our data center that we have recently deployed. We have been using Cat3560s as our top of rack switches in the data center and now wish to refresh them. What switch model would be best for this? Here are our requirements:
1 RU in size
48 copper ethernet ports
At least two 10Gb SFP ports for multimode fiber back to the N7K VPC
VTP client mode (our N7K acts as a VTP server)
Non-blocking switching performance
We had focused on Nexus 3000 series switches but they do not support VTP client mode.
Thanks!
Jeff
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05-19-2014
03:31 PM
Hi guys, I am implementing IPv6 on a lab environment. My ASA is running 8.2(5). I am mapping the outside IP address 66.181.12.242 to 192.168.22.101 with the following command: static (inside,outside) 66.181.12.242 192.168.22.101 netmask 255.255.255.255 I have given the inside host an IPv6 address of ABCD:192:168:22::101. I want the ASA to translate ABCD:66:181:12::242 to the internal ABCD:192:168:22::101. What is the equivalent command to do this in IPv6? FYI, this is just a lab environment so the outside IP address isn't actually on the Internet. For the purposes of this lab I need to translate IPv6. Thanks! Jeff
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10-01-2013
09:00 AM
We have a requirement to have CUCM challenge for authentication when it receives an invite from a SIP 3rd Party Device. Apparently, this functionality is available on other PBXs. How do we configure it on CUCM? Thanks! Jeff
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Created by
jmunoz19
in VPN and AnyConnect
in VPN and AnyConnect
08-16-2011
01:40 PM
08-16-2011
01:40 PM
IP Communicator supports 64 bit OSes. I'm running CIPC 8.6 on 64 bit Windows 7. Thanks, Jeff
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Created by
jmunoz19
in Contact Center
in Contact Center
05-24-2011
11:24 AM
05-24-2011
11:24 AM
You could setup a Single Number Reach DN in CUCM that is dedicated to each person's cell phone. When that DN is called, it'll ring the Remote Destination as well (cell phone). The Remote Destination "Answer Too Soon" timer ensures that if the call is answered too quickly on the cell phone (by going straight to voicemail) CUCM will pull the call back into the system and instead send the call to Unity Connection. Setup that DN on Unity Connection to foward back to an escalation application on UCCX. The escalation application could be a call redirect to the manager's cell phone, for example.
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Created by
jmunoz19
in IP Telephony and Phones
in IP Telephony and Phones
04-14-2011
02:20 PM
04-14-2011
02:20 PM
Does anyone have Visio icons/stencils for 9951/9971 and 7985 video phones? I can't find any on CCO. Thanks! Jeff
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04-14-2011
02:06 PM
Can the ASA reach the internal IP addresses? If they are not directly connected to the ASA's inside network, does the ASA have a route on the inside interface to get to these addresses? Otherwise, it'll want to go out the default route.
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04-14-2011
12:47 PM
I'm kind of confused as to what you are asking. Having a /30 between the ASA and the provider and then a different /26 network for static entries should not be a problem. It sounds like you are not setting up your static entry correctly. It should look like this: static (inside,outside) xxx.yyy.zzz.65 [INTERNAL IP] access-list ACLNAME extended permit tcp any host [INTERNAL IP] eq https The fact that your /30 and /26 are different ranges does not matter.
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Created by
jmunoz19
in Contact Center
in Contact Center
04-05-2011
01:28 PM
04-05-2011
01:28 PM
I believe this won't be a problem for you since you're not even using the CUCM subscribers in question. However, if I were you I would take a maintenance window and bounce the UCCX services anyway to make sure there won't be lingering issues.
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04-05-2011
01:19 PM
No, you don't particularly need layer 3 switches for VoIP. You would want to setup a voice VLAN that's dedicated to the phones but it wouldn't really be different from any other VLAN in the environment. Whatever is doing layer 3 now can do the routing for the new voice VLAN.
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Created by
jmunoz19
in IP Telephony and Phones
in IP Telephony and Phones
03-31-2011
07:46 AM
03-31-2011
07:46 AM
Try this: Create a user with a username and password of the extension. Create a Third Party SIP Advanced device. Associate the username as the digest user. Create a DN with the extension number.
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03-21-2011
06:49 PM
This is a problem I've run into before. Do you have h.323 dial-peers on your CME router? If so, it'll accept calls from any Internet address. You either need to put access lists on the outside interface of the router or put a firewall between CME and the internet.
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Created by
jmunoz19
in IP Telephony and Phones
in IP Telephony and Phones
03-09-2011
08:11 AM
03-09-2011
08:11 AM
It sounds like the username and password on the Polycom might not match the username and digest password on the user in CUCM. Typically I make the username the same as the DN and the digest password the same as the DN as well. You set the digest password in "Digest Credentials" field under the user.
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03-08-2011
10:31 AM
It sounds like you're looking for the "Forward No Answer Timer" under "Cisco CallManager" service parameters. The timer is in seconds, not rings. Typically I raise this timer up to 18 seconds when deploying new clusters.
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03-08-2011
10:00 AM
It sounds like the ASA does not have a route to the inside LAN network. When you ping from the LAN, the return packets don't have a route back so they follow the default route to the Internet. You need to add a route for the LAN network pointing to the router.
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Public Statistics
| Date Registered | 01-25-2009 04:03 PM |
| Date Last Visited | 03-14-2018 12:24 AM |
| Total Messages Posted | 146 |
| Total Helpful Votes Received | 54 |
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