IP Telephony and Phones

I have a CUBE deployment Cisco 2911 15.41T1 running fine, sip trunks to UCM 10.5.1 and to the SIP provider, again running fine. The designers of this project specified that a 2-port FXO is installed in this cube to support a Bogen TAMB unit, not the ...

How to learn the provisioning of a new voice circuit from local carrier. where is your dial tone coming from? T1 to gateway to router? Establish CSS, partition, gateways, translation patterns?New to call manager cheers 

All, Customer doesn't want to have "List is in directory" option on the welcome greeting. Is there a way to remove it??They don't want customer an option to choose the option. Any help is much appreciated.. Thankks

RTMT - Real Time Data Is there anyway to download these files so you can look at and search through more then 500 records at a time? I did search but my googling skills are failing me.  Thanks!

Hi,i have a couple DX 650s int my organization, but i cant seem to get Paging to work on them.i am running CUCM 9 and InformaCast 8.5. any guidance will be appreciated. thanks.

Hi all,  please help me with this MoH issue.Problem: MoH will not play at the branch office.  Callers from across the PSTN get the silent treatment during hold.Central site:CUCM cluster v7.1MoH Server is using 239.2.1.1 / port 16384 with "increment m...

Hi Guys,I have an issue for one of my customers.I have been carrying out an upgrade for this customer. The upgrade was done from CUCM 6.1.5 to CUCM 9.1 with a change in hardware from MCS to UCS.Further to the backup and restore from MCS to UCS, I did...

Is Disconnect Cause Code 172 documented anywhere? I cannot find anything. It would be nice to see the description of the cause code!Thanks!

Hello to everybodyI have SIP trunk beetween CUCM and cisco 2901 with CME and our customer want to see the names on the phone displays (on CUCM) instead of phone numbers. As I understood I should use SIP normalisation script to do this. I don't have a...

Hi, we are using a "0" as a prefix to dialout to the PSTN (ISDN PRI) on our CME router (2921). The leading 0 will be stripped off  on the dial peer via a voice translation rule: voice translation-rule 11 rule 1 /^00\(.*\)/ /0\1/voice translation-prof...

Anyone know where I can find free training for Cisco Call Manager 10? Cheers