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- About kelvin.blair
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Member since
07-30-2004
572
Posts
401
Helpful
35
Solutions
Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
01-20-2014
08:43 AM
01-20-2014
08:43 AM
Glad I was able to help!!!
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
01-20-2014
08:43 AM
01-20-2014
08:43 AM
+1 on Java response. I've seen this multiple times. The solution is to reboot the entire CUCM cluster which usually resolves the issue. However, i've had one time where it required root access to modify and correct the issue. Its only be one time and I can't remember which version it was. I know that it was caused by a bug in the system. Thanks!
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
01-20-2014
08:39 AM
01-20-2014
08:39 AM
Can you provide us a little more detail to your setup? CUCM or CME? What Version? How are your gateways setup? H323 or MGCP What is your PSTN connectivity? POTS, PRI, SIP Thanks!
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
01-20-2014
08:27 AM
01-20-2014
08:27 AM
If you are using Cisco Unity Connection, then you can use the system schedules to accomplish your question. Basically, you will tell option 0 to send to a call handler "Called Operator" or whatever. you will tie a schedule to this and use the transfer rules Closed & STandard. During Standard Hours (Business Hours) transfer to X and During Closed Transfer to X. Another way is to route the call back to CUCM and use ToD routing in CUCM. The easiest is to use Unity.
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01-14-2014
09:48 AM
Leanardo suggested IP Routing which is one thing to check. The next thing to check is the sip inspection rules. Doing SIP Inpection on an ASA for this type of design can cause you issues.. So you might want to turn that off in the ASA as well.
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
01-13-2014
06:49 AM
01-13-2014
06:49 AM
The feature your are referring to is called "Jabber Guest". Here is the link that talks about Jabber Guest. http://www.cisco.com/en/US/products/ps13410/index.html Thanks, KB
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-04-2013
05:59 AM
10-04-2013
05:59 AM
Make sure ANN_1 and ANN_2 is assigend to a MRG. Make sure the MRG is assiged to a MRGL and it needs to be associated to both the SIP trunk coming inbound to the CUCM server as well as the VoicePorts configured for Unity Connection. Generally assigning the MRGL to the device pool that is used for Voicemail will work. This has to be done.
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-04-2013
05:54 AM
10-04-2013
05:54 AM
So you get no ring back when transfering from Unity back to CUCM?
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-04-2013
05:49 AM
10-04-2013
05:49 AM
How is connection integrated? SIP or SCCP?
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
12:23 PM
10-03-2013
12:23 PM
On the trunk you can assigned a Media Resource Group list and in this list is a Media Resource Group that contains certain resources like Conference, MTP, MOH and ANN. ANN=Annunciators. MRGL can be assigned at the trunk level in this case or device pool. My question has a MRGL been assigned to either place and if so does it contain the CUCM Annunciators resources?
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
09:40 AM
10-03-2013
09:40 AM
Do you have an MRGL assigned to the trunk that contains a ANN..? Also, bind you sip traffic to a single interface. this can cause issues as well. For example: voice service voip sip bind control source-interface GigabitEthernetx/x bind media source-interface GigabitEthernetx/x
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
08:44 AM
10-03-2013
08:44 AM
Going out on a limb considering i can't look at your CUCM trunk configuration.. On your SIP profile in CUCM, do you have disable early media 180 checked? try checking this to see if it helps or unchecking it. One thing that is really bothering me as well is this statement on the dialpeer to CUCM.. By default CUCM is delayed offer. voice-class sip early-offer forced is not needed to CUCM. It wasn't until 9.X that early offer is supported, but you need Early Offer support for voice and video calls (insert MTP if needed) enabled to do otherwise you do it at the cube level as you are doing for outbound calls..
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
07:46 AM
10-03-2013
07:46 AM
First Suggestion: Your matching incoming dialpeer of 10 and outgoing dialpeer of 2. Dialpeer 2 add incoming called XXXXXX <<<--should match destination-pattern progress_ind setup enable 3 progress_ind progress enable 8 see if that helps.. provide debugs again if no go..
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
06:53 AM
10-03-2013
06:53 AM
Try using the commands I suggested on your incoming dialpeer. If that is unsuccessful, please provide a sho run excluding any passwords. Also, please provide the following debug's: debug ccsip messages debug ccsip calls debug voip dialpeer inout Thanks,
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Created by
kelvin.blair
in IP Telephony and Phones
in IP Telephony and Phones
10-03-2013
06:09 AM
10-03-2013
06:09 AM
What does the cube dialpeers look like? You might be needing this on the voip dialpeer, but without seeing your dialpeer configuration for these calls it might be hard to tell. You can try this and see if it works. progress_ind setup enable 3 progress_ind progress enable 8
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01-22-2015
There are tons of products out there that you can use.. Here are a
few:Solarwinds CDR trackerSplunk ...
01-22-2015
Can you run the following debugs:debug voip dialpeer inoutdebug voip
ccapi inoutMake a test call and...
09-24-2014
The Cisco 7975 phone is a SCCP phone and colored. The 9971 phone is a
sip phone also colored has blu...
07-15-2014
Take a look at the following write up from Mr. Bell. This should answer
your question.
http://www.ne...
07-15-2014
Can you also add "show ccm-manager" and run "debug voip ccapi inout",
"debug voip dialpeer inout" al...
03-19-2014
Ali, CUCM does not have a way to password protect a meetme conference.
However, there are a few work...
01-20-2014
+1 on Java response. I've seen this multiple times. The solution is to
reboot the entire CUCM cluste...
01-20-2014
Can you provide us a little more detail to your setup?CUCM or CME?What
Version?How are your gateways...
01-20-2014
If you are using Cisco Unity Connection, then you can use the system
schedules to accomplish your qu...
01-14-2014
Leanardo suggested IP Routing which is one thing to check. The next
thing to check is the sip inspec...
01-13-2014
The feature your are referring to is called "Jabber Guest". Here is the
link that talks about Jabber...
Created by
kelvin.blair
in IP Telephony and Phones
10-04-2013
10-04-2013
Make sure ANN_1 and ANN_2 is assigend to a MRG. Make sure the MRG is
assiged to a MRGL and it needs ...
Created by
kelvin.blair
in IP Telephony and Phones
10-04-2013
10-04-2013
So you get no ring back when transfering from Unity back to CUCM?
Created by
kelvin.blair
in IP Telephony and Phones
10-04-2013
10-04-2013
How is connection integrated? SIP or SCCP?
Public Statistics
| Date Registered | 07-30-2004 08:25 AM |
| Date Last Visited | 11-19-2019 08:56 AM |
| Total Messages Posted | 572 |
| Total Helpful Votes Received | 401 |
Helpful Votes Given To
| User | Helpful Count |
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| 4 | |
| 5 | |
| 4 | |
| 5 | |
| 5 |
.jpg "Hall of Fame Master"){kind=icon}
.jpg "VIP Mentor"){kind=icon}
Helpful Votes From
| User | Helpful Count |
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| 10 | |
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| 4 |
