IP Telephony and Phones

If we are active directory enabled with CCM 8.6.2su2 and we are using TAPS/BAT to import phones, can we associate the users with the phones as the owner,  and under the line associate the user, and in end users set their directory number as their pri...

Is there a way to pull a report made on all FXS ports on a Voice Gateway? We are attempting to determine which analog phone could be calling 911. Be it accidental or on purpose we need to determine the culprit so to curb this action.

Hello,We have upgraded our CUCM cluster from 8.5 to 8.6 on UCS (VMWARE).When I look at the VMWare disks I see the following:Publisher:1x 80 GB disk Subscriber:2x 80GB diskSubscriber:2x 80GB diskIt looks like the refresh upgrade has coused an issue on...

Hi Cisco lovers,Got a 1861E router with voicemail licenses. The main part number is "C1861E-SRST-F/K9" with a description "1861E, 8-user SRST or CME, 4FXS, 4FXO, 8xPOE, SP Svcs, HWIC slot". According to data sheet http://www.cisco.com/en/US/prod/coll...

Hi all, is there a possibility to change on CUCM 8.6 the ring volume per tone? I´ve created three lines on one phone and my goal is to get a ring tone just for the first line, for the other two lines the phone should just flash, not ring. Did anybody...

I'm running CUCM 8.6.2 in non-secure mode, and I've been getting certificate expiry RTMT/Syslog alerts. Not a problem, I thougt. Easy. Untl I started looking into it...Exactly what are all the certificates for ? And which ones have to be done out-of-...

We got a problem, which our system is using CUCM 6.1.4, and we using 7940 cisco ip phone with firmware sccp 8.1.2, we also using EM for login and logout for user, and the problem is, when the user try to logout after login to the ip phone, the user g...

Hi guys, I am looking for some advice regarding a proof of concept setup we are test lab-ing before we commit to getting a CUBE.Please see attachment for our test lab setup..We have created a SIP trunk from our CUCM7.1.5 to another PBX ( Asterisk 2.6...

I have a quick question (hopefully) regarding an ATA 187 and secondary DTMF tones. Our issue deals with a new ATA 187 (SIP) and calls placed from the device into MeetingPlace Express or AT&T Teleconferencing Systems. We are able to dial the number to...

I have a cisco 2851 switch where my 8 telephone lines (RJ-11) comes into it and it converts them into IP (RG-45) goes outNow I have a fiber optic cable from the ministry of communication where my phone lines are configured, how can I convert the fibe...

Does the Cisco 2851 router accept a LC connector?The one I got only have RG-11 connectors where I input the phone lines from the ministry into it and then output them as RG-45 ethernet cables to my IP Phones.As now the ministry is providing fiber opt...

Hi all,This is my first question in this forum, so please don't be to hard on me :-):I have installed two new DHCP servers in our German branch. My laptops are using the new server like they should (if I disable the old DHCP server service, they conn...

I have a 7961 over a DMVPN tunnel and I have factory reset the phone.  It now will not upgrade.  I'm on a distant site and the call manager can see the MAC addy of the set.  My question is this:  The tunnel is very unstable, going down multiple times...

Hello, I am learning the basics of SIP.I have configured CME with an outgoing dial peer as follows:dial-peer voice 1 pots destination-pattern 0[2-9]....... session protocol sipv2 session target dns:xxxxxxxxx voice-class codec 1  no vad!Everything is...