Hello, We are currently using a Be6000 system and when we schedule a call forwarding on our phones (model 7945) to external numbers, it works perfectly when the original call comes from an internal position, but when it is an external call then this ...
Hi all, I have a Voice gateway Cisco 4331 in which already 7 T1s are configured with high preference dial-peers for PSTN calls.My new requirement is to add 2 DID ranges 1XXX & 7XXX, I will configure these T1s on new interfaces. Now if anyone from a...
Site is running CUCM 11.5.1 Cluster (1pub 2 sub), VG350 for Analog Connections, Dual ISR 4431 Voice Gateways out to PSTN via MGCP. Once a call is set up, Call managers Step out of the Communications stream and RTP/RTCP is endpoint to endpoint only,...
Hi all. I'm trying to understand how I can get phones that aren't able to route to one another directly to talk through a DMZ, most likely using an MTP. We have a standard production network in which our CUCM cluster, PSTN gateways, and phones are ...
I have replaced some older model phones with the 8841 series phones. The older phones had the iDivert feature as a softkey. I am trying to create an iDivert softkey to mirror what these users had previously on the 8841. I have managed to get a "Decli...
I have had to restart one of our two 11.5 CUC cluster servers twice in the past 6 months due to power/environmental issues. Both times, It comes right up, BUT the Administration account password no longer works on either server, and I end up going i...
Can someone tell me if it is possible to not show the call history on a phone when the handset is picked up. We have a bunch of different model phones, 9965, 8845, etc. When someone picks up their handset it will show all the call history of the ph...
Is it possible to terminate a Conference Now session after the Host disconnects from the call? I have been reading numerous discussions on Conference Now and it doesn't look to be configurable like it is for the Ad Hoc conference - using the Drop ...
Hello Community, I am trying to understand how to apply a Caller name so all calls to specific number we have appear as a caller name not a number on the phones. Would we simply just apply this on the line or does it need to be done elsewhere? I ...
Hello, I have a problem when, from a SCCP phone I enable the call-forward all to another SIP extension (only SIP extension), the error in debug log is "Cause i = 0x80AF - Resource unavailable, unspecified.". This happen both from internal and extern...
i have a unity 11.5 , i have created a AA and call handler which during the greeting if you press a number you get transferred to an extension and set it to "release to switch" , the problem the call disconnect after three rings? how can i increase t...
Hi Guys, I'm currently upgrading an Bank Market room, going from the Old IP-Trade turrets on an Cisco cucm to Cisco 8851 + KEM. Cheaper to buy and maintain maintenance. The Turrets however have an snapshot add-in to playback recordings from prev...
Hello, I just subscribed to a 1-line SIP discovery offer to test the compatibility of the VOIP hardware that I have in my possession, this is a Cisco 7970G VOIP phone with SIP70 version "SIP70.8-5 -3S ". I realized all the configuration of the phone ...
We are running a cluster with 2 CUC server with version 8.6 on them, after a recent reboot of both server we were having replication issues and I was able to solve the problem following some steps from a different forum post about dbreplication reset...
Hello,I just registered a 3rd Party SIP Phone on CUCM 9.1, i can receive calls but im having problems to dial from those phones.Attached is a trace from CUCM.. Thanks
I have 1 number that users call in on and have 2 backend servers that connect via SIP Trunks. I would like the calls to rotate between two servers so that they are used equally. First I created a Translation rule to route to a Hunt Pilot/List that...
