IP Telephony and Phones

Can anyone please send me a link for configuring trunk group to allow each DDIs on each E1 channel? I believe i saw this comment somewhere that the following dummy pots and voip is needed as well but cannot find that as well dial-peer voice 1 pots in...

what is the difference between CP-BATT-7925G-STD= and CP-BATT-7925G-EXT= ?  

Hello All,For a customer in a high security environment we're installing 3905 phones for public usage. The problem is that these phones still display a "phone information" menu option which contains ip addresses for the phone and the active callmanag...

There is a problem with the off-hook switch on 7940/7960 phones. After some time the switch does not work. When you pick up the handset nothing happens.What is the best way to fix this?1: Clean the switch circuit board with alcohol?2: Replace the swi...

Current setup: CUCM ver 8.5The CUCM is integrated with LYNC using SNR. Each extension is configured with remote destination & pointed toward Lync using SIP. eg : extn : 1000 RD : abc@company.com. When someone calls 1000 user can either pick from phon...

Hello,I am unable to make Video call between Head Office and  branch office in Cisco IP phone 8945. Two branches are connected through Inter-Cluster Trunk (Non-Gatekeeper Controlled) Head Office details:CUCM version 10.5IP Phone 8945 Active load SCCP...

Hi,I m trying to find solution for the following problem that I have to solve in our work environment :1- I configured my router as a CME (cisco 2911), I ve 10 ephone-dn dual line (1XXX), and 10 ephones (cisco 6921).2- Supposes we have a tem composed...

Hi, i have setup Cisco 2811 as voice gateway with call manager 7.1.3 having auto attendant on unity connection. when someone calls to landline, auto attandant response and works fine if call flow reach to extension and i attend but analog line remain...

Hi guys. I have calls running between 2 voip systems trough sip and i have about 7 hop between them (2 mpls and the rest of them internal between routers (fiber connection) ) .When I see logs I see too big jitter from 1000-4000 ms , packet loss 15%-4...

Hello Team,, We are facing the issue with Cisco CME with call manager .If we are trying the SIP trunk calls  from call manager not going to cme ,from cme to call manager sip call is connecting but there is no audio.with the same setup H.323 calls goi...