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- About scooter817
09-19-2019
Stats
Member since
04-01-2008
610
Posts
5
Helpful
2
Solutions
09-11-2019
06:42 AM
Piyush I made a mistake when I copied the dial peer information into the post, but i got the dial-peer created yesterday afternoon. This is what was created and now since I have that done I'm going to start reading on how to configure dial peers. I'll going to take my time so I can get a good understanding. voice class server-group 1 ipv4 192.168.248.3 pref 1 ipv4 192.168.248.1 pref 2 dial-peer voice 120 voip description Incoming PRI calls to CUCM preference 1 destination-pattern 6503[6,9]5....$ progress_ind setup enable 3 session protocol sipv2 session server-group 1 voice-class codec 10 voice-class sip early-offer forced dtmf-relay rtp-nte no vad
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09-09-2019
01:58 PM
Jamie I looked over the configuration manual and I wanted to know can you tell me if this is correct for calls coming into CUCM. The numbers that I'm working with are in the 650365XXXX and 650395XXXX range. dial-peer voice 100 voip description Incoming calls to CUCM1 preference 1 destination-pattern 650365.... session protocol sipv2 session target ipv4:192.168.248.3 voice-class codec 10 voice-class sip early-offer forced dtmf-relay rtp-nte no vad dial-peer voice 101 voip description Incoming calls to CUCM2 preference 1 destination-pattern 650365.... session protocol sipv2 session target ipv4:192.168.248.7 voice-class codec 10 voice-class sip early-offer forced dtmf-relay rtp-nte no vad dial-peer voice 102 voip description Incoming calls to CUCM1 preference 1 destination-pattern 650395.... session protocol sipv2 session target ipv4:192.168.248.3 voice-class codec 10 voice-class sip early-offer forced dtmf-relay rtp-nte no vad dial-peer voice 103 voip description Incoming calls to CUCM2 preference 1 destination-pattern 650395.... session protocol sipv2 session target ipv4:192.168.248.7 voice-class codec 10 voice-class sip early-offer forced dtmf-relay rtp-nte no v
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09-09-2019
08:43 AM
I have a new CUCM cluster that I'm taking over for our California region and they are porting some numbers from AT&T to their Level 3 SIP trunk. I want to make sure that when the port happens that the new numbers are able to make calls outgoing and incoming. I spoke with TAC and they told me that from looking at my route patterns the outgoing will work fine. But he informed me that he would have to engage someone from the gateway team to look over my dial peers to configure the correct one for the new numbers to work for incoming. The engineer is currently on vacation this week and the port is happening this Friday. I wanted to know can someone look over these dial-peers and tell me what do I need to add if anything to make these new numbers work for incoming calls. The numbers that I'm speaking of are in the 650365XXXX and 650395XXXX range and thank you for the help and I look forward to your replies. Eric
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Created by
scooter817
in IP Telephony and Phones
in IP Telephony and Phones
08-30-2019
07:52 AM
08-30-2019
07:52 AM
Recently I've been getting more and more people complaining about calls disconnecting. The users are saying that they will be on a call from about 10min to 30min and the calls just drops and they get a fast busy. I wanted to know has anyone seen this before and thanks in advance for the help. Eric
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06-28-2019
09:33 AM
Hi Everyone I had a strange issue that has been happening to users and today it happened to me as well. I had a call today at 8:30am and the call went on until 9:15 and then it just dropped. So I looked back in my call history and the call isn't there, the users who came to me told me the samething. They would be on a call and after a about 20-30 min the call would just drop and it wouldn't show up in the call history. Any help would be appreciated and I look forward to your replies and have a great weekend. Eric
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06-26-2019
07:21 AM
Hi Everyone I have an issue that I'v been working on with TAC and Level 3 for about 2 weeks and I really need to get this fixed for this user. I have an IVR that I setup for him and everything works fine except when users press on option like say 1, they get a small piece of MoH about 2 seconds and then dead air until the other end picks up, they aren't getting any ring back.I'm going to attach the log files from the CUBE and if someone can look at this and tell me what they think is causing this I would greatly appreciate it and thanks in advance. From CUBE perspective: CUBE incoming: Call-ID: 1461716728_71036257@8.45.217.85 CUBE outgoing(transferred): Call-ID: 8BBA7387-96B411E9-AF74D144-F6C3A48A@172.16.100.13 Calling Number=(469)203-9813 Called Number=(817)287-2075
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03-01-2019
11:15 AM
Here you go
https://www.cisco.com/c/en/us/support/customer-collaboration/unified-contact-center-express/tsd-products-support-series-home.html
https://www.cisco.com/c/en/us/support/customer-collaboration/unified-workforce-optimization-call-recording-11-0-1/model.html
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02-28-2019
02:04 PM
Thanks Anthony, were upgrading our entire UC suite to 12 at the end of the year so then we can use both links to point to different servers.
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02-28-2019
10:10 AM
Hi Everyone
I had an issue yesterday where my primary node went down and users who weren't logged in couldn't access the Finesse web page when they came back from lunch. I didn't know that when the primary node goes down that users who aren't logged in aren't automatically failed over to the second sever. Anthony explained that the easiest way is to publish links to access Finesse from both servers. So of course the primary is back up now and the link for Finesse works fine, but when I copy and past the link for the second node it keeps redirecting me back to the link for the primary node. Will the secondary link only work if the primary link is broken or is there something else that I need to do on the second node to make the link work.
Eric
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- Tags:
- uccx
Labels:
02-28-2019
08:34 AM
Thanks Anthony, I had no Idea that if you aren't logged into Finesse and the primary node goes down it wouldn't automatically fail you over when you try and log in.
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02-27-2019
02:58 PM
Shalid
From reading the information it seems like the only way to get users to be able to log into Finesse is to have both URL published on a load balancer.
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02-27-2019
01:29 PM
I have a quick question, we have redundant UCCX servers and a few moments ago the master went down and no one could log into Finesse. Shouldn't the slave have taken over and allow the user to still log into Finesse, my manager asked me why didn't this happen and I wanted to find out for him.
Eric
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- Tags:
- uccx
Labels:
11-16-2018
07:11 AM
Hey Chris
I actually opened up a case with TAC and the engineer on the phone helped me configure the script the get done what the help desk manager needs.
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09-19-2019
Hi EveryoneI have a group of users who are in a hunt group and they want
to have a night service ena...
09-11-2019
PiyushI made a mistake when I copied the dial peer information into the
post, but i got the dial-pee...
09-09-2019
JamieI looked over the configuration manual and I wanted to know can you
tell me if this is correct ...
09-09-2019
I have a new CUCM cluster that I'm taking over for our California region
and they are porting some n...
09-03-2019
Hi EveryoneI currently manage our CUCM system and we have a totally
separate system in our Oregon re...
Created by
scooter817
in IP Telephony and Phones
08-30-2019
08-30-2019
Recently I've been getting more and more people complaining about calls
disconnecting. The users are...
06-28-2019
Hi EveryoneI had a strange issue that has been happening to users and
today it happened to me as wel...
06-26-2019
Hi EveryoneI have an issue that I'v been working on with TAC and Level 3
for about 2 weeks and I rea...
05-23-2019
Hi EveryoneI wanted to know if someone can tell me what do these
acronyms mean when it comes to lice...
Public Statistics
| Date Registered | 04-01-2008 08:33 AM |
| Date Last Visited | 09-19-2019 09:34 AM |
| Total Messages Posted | 610 |
| Total Helpful Votes Received | 5 |
Helpful Votes Given To
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