I'm having trouble trying to figure out the best way to route calls to agents from a caller input, then send to a higher skilled agent based on that input. For example: Prompt: For Customer Service press 1, for Warranty, Press 2, for All other inquiries press 3. Inputs: 1. Route to agents with Minimum skill level of 5 2. Route to agents with minimum skill level of 9 3. Route to agents with minimum skill level of 2 I want higher skilled agents to be able to field calls in a lesser skilled group as well. What's the best way to accomplish this? Thanks in advance....            ... View more

I have a UC320W setup with no auto-attendant, and 5 phones that share two FXO lines for inbound and outbound. It seems that if you answer from any phone on either line during the second ring, the call drops. If you answer between rings, after the first or after the second, it answers fine. Has anyone seen this issue before? thanks, Marc            ... View more

Hey All, I have read in the Cisco SRND for UC that you Microsoft Windows 2008 R2 64 bit Active Directory servers are not supported by Cisco for LDAP users. Can anyone confirm that this still works with CUCM 8.x? It not being supported is one thing, but it not working would completely suck. Thanks Marc ... View more

Hello I have an environment that is running Cisco CUCM 7.1.5 and has SIP trunks to two different Avaya PBXs (Option 81). The sound quality from a Cisco phone to a PBX based phone is good, however the sound quality when the call originates from the PBX based phone to the Cisco phones is terrible. I've started to collect SIP traces over the trunks to run them through SIP Workbench and look for clues. Does anyone else have any good tips for this? thanks MP ... View more

Hello all I'm running Call Manager 7.1.5 and i'm trying to obtain packet statistics for some SIP trunks that we have between the CM and some Avaya PBXs. I have already ensure that CDR only is disabled and I get IP phone to IP phone statistics, but I really need the stats for the SIP trunks. Does anyone know how I can obtain these numbers? Thanks! MP ... View more

hello I have several users that are using phones over asa phone proxy (ASA v8.2) and when they make calls from their phone to phones that are not on the proxy, two way audio is established. If a phone proxy user calls the phone of another phone proxy user, the call results in one way audio from the originating caller. so on the originating side, the user cannot hear the called party and also the originating caller has no recvr packets. any ideas? i've triple checked the MTP config on the ASA phone proxy and it looks fine. ... View more

Has anyone successfully registered a Polycom IP 4000 soundstation with Communication Manager 7.1.2 or any other version for that matter? I've followed the "Configuring 3rd party SIP devices" documentation located here: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b09sip3p.html#wp1011004 but have yet to get this working. Basically the phone won't register. thoughts? Thanks Marc ... View more

I have an AA running on a 2811. When you call from PSTN to the AA and use DTMF to reach a user, the phone rings fine and the call can be established, however as soon as the call gets forwarded to the voicemail that is located on a unity connection machine off site the call goes silent and nothing happens. We are using AAR and having all Voicemail travel over pstn. Any thoughts? transcoding problem perhaps? ... View more

I'm working with a customer that is using various ISRs (881, 1861 and 28xx) with the following services: DMVPN, ZBFW, QoS and Voice. The max throughput we've seen on the 1861s is 8mbps and on the 2801s it's less than 20mbps. Is this normal and acceptable? ... View more