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Hi, I was wondering if anyone has used PCP to sync with LDAP to import a global directory which sync's to CUCM?The issue I have is when I sync PCP to LDAP and CUCM to PCP this works fine for contacts at the site which I want to provision services to ...
Hi there,I have an install of a CME with FXO ports going live this week and I was wondering if someone could check over my sound extract and confirm for me that the frequency is 415 and cadence is 510 280. I can't test until the lines go over but jus...
Hi,I currently have an issue connecting my FXO point into a telco phone line (Linksys SPA8000 box), whereby any incoming or outgoing calls are immediatly disconnected. An incoming call will ring and once answered the call will immediately disconnect....
Hi I was wondering if anybody knew how to make custom text appear on the agents phone when it is ringing rather than just the Cti port number. I have the enterprise call info working and have it showing on the cad software but would like it to appear...
Hi, I am in the process of going through a new UCCX fresh installation and when I try to run the CAD Client Configuration.msi it is showing the following error:Error opening DB: error = d<110>Anyone seen this one before?ThanksCarl
I'm having this same issue however the UDS url is showing the wrong hostname (it is missing one of the characters)
It is not an issue with my DNS records they are correct:
SRV records ok
DNS A records ok
hostname set on CUCM (System>Servers) are corr...
Hey David, You're right it's not really a solution for the real world but may get you out of jail for a key conference.I've logged it with my account team and with an SE I know plus will put it on a spark room im in with some Cisco collab guys so hop...
Hi, I have this same issue too and went through the process with tac. You can mute video through TMS if you need to manually mute each audio dial in participant and it removes the avatar but you'll need to do this each time there is a conference whic...
An 8831 will be fine using fast track, try the following config (make sure you have the usual sip config in also): voice register pool-type 8831 xml-config maxNumCalls 2 xml-config busyTrigger 1 xml-config custom sshAccess 0 num-lines 2 description ...