Do you get any information at all? some commands only work for certain protocols, if you have an MGCP deployment I don't know if the info is there, also POTS vs SIP the commands are different try with something like:   show voice call status show isdn active   Cheers! ... View more

Not anymore, since the old model was retired a few years ago but the phone we have are CP-8821-K9-BUN ... View more

Extension #10 and #11 is the same (104), that could be a problem. More information is needed, but you should check that you do not have duplicate extensions and that all the extensions are assigned to the correct phones. I'm assuming all the communication is local, but if they are not, you will need to consider call routing also (dial-peer and/or SIP trunks)   Hope this helps. ... View more

We had troubles with older models of 8821 as soon as we upgraded our wireless, since those old phone only really work well on 802.11a/b which SUCK! After upgrading to 8821G it was pretty much plug and play and we do not have problems with them, but any wireless phone that supports SIP should be a feasible alternative.   Regards. Rolando A. Valenzuela ... View more

PLAR should be another option also, just look documentation online for either option. https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/212073-CUCM-PLAR-Configuration-Example.html ... View more

Just to make sure, is this accurate? Cisco Phone -> CUCM -> Gateway -> PSTN ---- This WORKS Third-party Phone -> CUCM-> Cisco Phone ---- This also WORKS   Third-party Phone -> CUCM -> Gateway -> PSTN ---- This does NOT work.   If it is, the problem is either between CUCM -> Gateway or Gateway -> PSTN, please check Supported codecs. Is the call being transcoded? Media Flow Through vs. Media Flow Around. ... View more

Hello Jonathan, What type of data are you looking? maybe you aren't reading output right? ... View more

I hope everybody found this link https://cway.cisco.com/tools/vccalc/ for call estimates Registration estimates are different, just consider the boot up process for an IP Phone and measure that.   Power-on self-test (POST) The phone begins the boot process. The phone uses Cisco Discovery Protocol to learn the voice VLAN. The phone initializes a basic IP stack. The IP phone (DHCP client) sends a DHCPDISCOVER request to the 255.255.255.255 broadcast address. A DHCP server (sometimes it is CUCM) returns a DHCPOFFER message and assigns the following for the requested scope: a free IP address, the subnet mask, the default gateway, the DNS server (optional), and the TFTP server (option 150) for the scope. This information is sent to the DHCP client (the IP phone) using the broadcast address 255.255.255.255 (the router uses the IP phone MAC address at Layer 2). The IP phone takes the values received from the DHCP response and applies them to the IP stack of the IP phone. The IP phone uses the value received in option 150 to attempt to retrieve a configuration file from the TFTP server. (Usually CUCM) Phone register with call manager. The file size (based on feature, dialplan, protocol and phone model) varies + all registration signaling messages will give you the bandwidth for a phone - It is something you will have to measure on your own most of the time. Actual calls will consume more bandwidth so as long as you have your math right for your QoS policies, you should not have problems with registration. ... View more

That's a concept error, you don't need an empty CSS to restrict calls, just don't assign one and it will have the same effect. What is the app name? I'm assuming device control is enabled and then the app is the one making the call, and not the phone, you will also need to check the hierarchy level, app CSS, line CSS, phone CSS, etc. ... View more

I was about to mentioned the same thing, configure your partitions/CSS correctly and try again https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/22325-part-css-tn.html ... View more

The call cannot find a matching dial-peer 1d02h: dp_delete_route_label_member_by_index:Existing dialpeer route label is null I'll suggest to revisit your dial-plan configuration, you should not combine incoming called-number, answer-address or destination-pattern Please refer to the following URL to understand the matching process:  https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html?referring_site=smartnavRD#topic3 ... View more

I won't create a full dial-plan for you, but a can give you some suggestions: If possible DO NOT uses overlapping extensions. If possible use E.164 format for all your extensions. You can use translations to allow the 4 digit dialing. Consider Toll-fraud. For a large deployment like this I'll say based gateway locations.   I don't have all the details but I'll do a modular partition/css design (more partitions can added depending on the needs)   global_toll_fraud region_emergency region_local region_long_distance region_international   Then you can arrange the CSS according to your needs specially on how to dial all the branches. Your diagram talks about HQ calling the branches, but what about the opposite? or even Region2-B1 calling Region7-B3? I'll suggests to do a calling party transformation so it will be easy to the other end just to press "redial" and call back.   Once you have all the extensions ready, the dial rules should be pretty simple since is just repetition for each branch - check if you can export all that in Bulk.   ... View more

The easiest way will be having a second for all the sales reps so they can do their job using this (maybe shared) Directory Number while having their personal line for other purposes. Another way to accomplish this is to move them to a different partition/CSS and do the masking on the Route Pattern.   I'll prefer shared lines just because it is easier to implement and troubleshoot avoiding a full reconfiguration of your system.   Rolando A. Valenzuela.     ... View more