IP Telephony and Phones

CUCM version 6.1.5Active directly version: exchange server 2010 SP1I did the following on RTMTStep 1:  Choose System > Tools > Alert > Config Email Server.The Mail Server Configuration window displays.Step 2:   In the Mail Server field, enter the e-m...

                   Hi ,Any one help me where i will get bmp2logo.exe tool to convert .bmp image to .logo formate,And plesae tell is tht requied to convert image type if i want to change logo in  7965G phones.

                can any one please advice    How long can the cucm sub can be disconnected or switched off without database loss ?

Do Cisco have any products that will do a details call accounting and generate reports for CME/CUE solution. Also for call recording, does Cisco have a solution for CME?If Cisco does not provide this, what third party call accounting and call recordi...

Hi,     I was wondering if there was something special reqiured to register mobile Jabber clients to the CUCMBE 3K, all the Doco I have read thus far seems to indicate that it is supported so I thought I would give it a shot as I have a spare enhance...

Hi,The other day we lost voice calls via our SIP Trunks. I have spoken with 3rd Party, although no Root Cause from them. I have some debugs, which seem to point to otherwise and wanted to know if I am reading them correctly.We believe a DNS issue was...

Hi I installed Cisco Jabber on Samsung Galaxy S3 for CUCME 8.6 ERROR "Please check your Internet Calling Settings and try again. If the problem persists, contact your system administrator" But it is  working fine with S2 and  other devices . Please h...

cmterm-7911_7906-sccp.9-1-1SR1.zip

There's a question https://supportforums.cisco.com/thread/2157114?tstart=30 where the poster has a problem firmware upgrading a phone. Doing a factory reset (123....) forces the phone to upgrade, but it still has problems.I'm guessing a CME config er...

         Hello Folks , we have two trunk one is sip and anther one is pstn , my voice mail is in some third country  and that is routed through SIP , now my requirent is if some body call at my extension from PSTN , it should 8-9 ring then disconnect...