11-04-2010 08:59 PM - edited 03-16-2019 01:45 AM
Is Unified Border Element licensing enforced on the 2900 and 3900 ISRs? If so, what command can I use to determine how many sessions my VGs are licensed for? A show version just says I have the UC license.
Thanks!
Jeff
Solved! Go to Solution.
11-05-2010 04:43 AM
CUBE licensing ("allow connections...." CLI) is not yet enforced (all honor-based at this time) although it will become so in future. For demo purposes an
eval license is required, and for production environments a CUBE license (FL-CUBE-XX) has to be ordered. All paper based and nothing to installl as such.
If u configure GK as well, u'd need to purchase and install GK license though.
11-05-2010 04:01 AM
CUBE is a RTU license and does not use a key. All you need to have is the UC license installed on the router to configure it.
http://www.cisco.com/en/US/prod/collateral/routers/ps10616/white_paper_c11_556985.html
Right to Use Feature Licenses
06-21-2013 12:30 AM
Hi ,
Is there any command to run on CUBE to find out how many sessions are active even though paper licence are applied or any violation is happening ?
regds,
aman
06-21-2013 01:21 AM
You can use "show voip rtp connections" to see the number of active calls.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
06-21-2013 02:31 AM
Hi AOK,
Can we see some sort of violation license as well?
regds,
aman
07-18-2013 11:16 AM
Hi
I already have a 3945 Router that have purchased long back
Now we are planning for a SIP TRUNK to Service provider ( PSTN).
We have UC License.
As we have not purchased CUBE license earlier with router , I would like to know
if its available to Purchase CUBE License and Session License
as spare ? if its Right to Use license..how to purchase and inegrate it with existing 3945 Router ?
==============================================================
02-16-2015 11:55 PM
Hey,
Can you share with me please the config of the CUBE ( dial-peer, sip configuration with PSTN etc).
Thanks a lot for help
02-17-2015 01:47 AM
Gabiulici
Here is a detailed guide for you.
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:<CUCM_Address>
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:<SIP_Trunk_IP_Address>:XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
11-05-2010 04:43 AM
CUBE licensing ("allow connections...." CLI) is not yet enforced (all honor-based at this time) although it will become so in future. For demo purposes an
eval license is required, and for production environments a CUBE license (FL-CUBE-XX) has to be ordered. All paper based and nothing to installl as such.
If u configure GK as well, u'd need to purchase and install GK license though.
06-20-2013 12:11 PM
Is this still the case?
06-20-2013 12:53 PM
Luis,
Yes it is still the same. CUBE licrenses are right to use, however you need UC license on the 2900/3900 gateways
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-12-2016 02:30 AM
Hi, this post on the licence being honor based was very useful.
Do you know if this is still the case nowadays? Also, I understand that the whole platform can be licensed so it will work up to the maximum allowable on that platform. How can you tell if your router (in my case a 2921) has such a licences already? I would hate to have to pay twice!
08-25-2017 02:16 AM
is there any way to check the Cube licenses?
08-25-2017 02:25 AM
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide