IP Telephony and Phones

We recently upgraded Call Manager from 8.6 to 10.5, and everything has been working well, with the exception of directory access.Currently, we have 6 different models of phones deployed in our network:696169416921390599518961 The last two (9951 and 8...

Hi All,   I have cisco call manager 10.5 with E1 4 digits & FXO ports , in addition to 5 branches in different locations connected to main site by sip trunk configuration ,every location or site configured with MGCP gateway and SRST to main site CM. ...

Hello, I wonder if it's possible to change the phone number that appears in call history  to be in relation with the number that appears on the screen when somebody calls. Indeed, with calling transformation pattern we can change calling number displ...

I have the CUCM 10 and i want to integration with ASTERIK, the ASTERIK is the Gateway for the another extension. I configure que SIP trunk fot the port 5060, for the PSTN Calls the Asterik is the Gateway. In the CUCM the SIP trunk is UP, but when dia...

Hi Forum, has anybody already integrated CUCM Release 11.x with MS Windows Server 2016 ADFS Role?   I´ve already had MS Windows 2008 (ADFS 2.0) and MS Windows 2012 (ADFS 3.0) successfully integrated, but can´t find any reference for MS Windows 2016 A...

Hello There, Please share what are the settings need to be done on CUC under System call handlers< Select AA< edit< Greetings? WHat does active phones settings mean if we go to Media Playback Recording and choose option phone to record Greeting?

Hi Team,I am new to this community but I am stuck in a strange situation.The calls from the Motorola TIG ( Telephone Interconnect Gateway ) to the Cisco EPABXconnected IP phone is absolutely ok and going as per the call duration restrictions of 8 min...

We have been investigating the FXO Bypasspair capability of VG350's where in case of a power outage the FXO bypass pairs switches their respective relays to ports 46 and 47 and provide dial-tone from the PSTN. However, is there a way to force this re...

Hi all, IPPHONE <-SIP-> UCM <-SIP-> CUBE <-SIP-> ITSP I try to enable interworking of KPML and INFO DTMF Methods. Our ITSP sends INFOs to us and the CUBE sends fine KPML NOTIFYs to UCM.But if I send KPML NOTiFYs to the CUBE, it does not generate IN...

I've been working here almost 2 years and I cannot for the life of me figure out how to change the display name on a new employees phone. We only have 4 people working in our office. This phone is WAY more than we need. I've read the manual, gone on-...

Hi, is there a simple way to give the users the opportunity to simply transfer the call to an internal DN after pick up the call via mobility. The following way is not simple to remember for the users: *82 (Exclusive Hold)new Call to EFAC DNUser PIN#...

CUCM - 9.1.2.11900-12Unity Connection - 9.1.2TT1.11900-2TT1IM&P - 9.1.1.41900-1Cisco 7970 SCCP I have had this happen twice now in the past month, so I am going to reach out for an answer.User picks up the handset to make or answer a call, but instea...

Hi. It is technically feasible to use 2 ISDN links from the same provider on a card with 1 E1 port.???? That is, having two trunks on an E1 card with an E1 port.   If feasible, how do I set dial-peers to route the call?   Thank you.

As a student radio station we are looking at intergrating our phone into our broadcast output, we currently have a IP Phone 7942.   we have tried using a headset adapter but when we plug into our external mixing deck, to connect to our speakers we do...