Check to see if your current firmware is the latest version before attempting to update. Download the latest version firmware which includes the Update Firmware Utility. To upload the firmware from your PC, see your Operations Guide.

Our Paging Server is a POE enabled, single SIP-endpoint enabling user defined paging zones through a multicasting connection to CyberData VoIP speakers.

The 99 zones supported are with a Bogen-type Zone Controller, in that our product is able to recognize the two-digit DTMF tone the user enters after it receives a cal and passes this tonal information to the Zone Controller so that the specific paging zone associated with the call can receive the audio.

Our IP paging products are programmed to traverse the NAT using Session Border Controllers (SBCs) of the VoIP hosting company or service provider. The SBCs act as an outbound proxy and manage the SIP traffic between the SIP server and the SIP endpoint behind the NAT.

Make sure your SIP phone is set to 101 for the DTMF payload type (Out of Band RFC2833).

This is a common problem when the re-registration time value is not set correctly. On our device, you need to make sure that the re-registration time value (in minutes) is less than that is set on the IP-PBX server.

In the PBX setup page for the extension of the CyberData device, find the Qualify= value and change it to NO. If the Qualify= value requires a numeric value, then change it to 0. Note that on some Asterisk systems (such as Intuitive Voice) this value is called the Heartbeat= value. Set the Heartbeat= value to NO, and then save the settings. Also, on the product's SIP Setup page, make sure that the Register Expiration (minutes) setting is set to less than 6 minutes (5 minutes is good) because it needs to be a value less than the Asterisk default value of 6 minutes. Save the settings after changing the Register Expiration (minutes) setting.

This is a bug specific to the CyberData Paging Server and occurs when videosupport is enabled on the PBX. We are working on a firmware fix. If you do not use VoIP Video Phones, then you can work around this bug by disabling videosupport. On the switchvox or asterisk system, in the SIP.CONF file [general] area, you'll find the following line: ?videosupport=yes? Either change it to no or delete it. You will probably have to restart the system for it to recognize the change. Most PBX servers have this as a selectable option.

There is a 3CX version 7.1 registration / timing bug. To correct this problem, complete the following steps: (1.) Log into the 3CX PBX system, and select SETTINGS -> ADVANCED -> CUSTOM PARAMETERS. (2.) Scroll down to select the NONCEEXPIRATION parameter. (3.) Change the NONCEEXPIRATION value from 1 to 2 and save the changes.

The power connection requires a +48v 380mA (min) power brick. The connector is 2.5mm with a positive center. Cisco's CP-PWR-CUBE-3 power brick is approved.

The cables or switch ports that you are connecting to are set in switch or hub mode instead of endpoint stations. The MDIX setting needs to be changed to MDI since our VoIP products are end stations. From the Cisco SLM 224P User Guide:

Change to MDI: MDI/MDIX displays the Media Dependent Interface (MDI) / Media Dependent Interface with Crossover (MDIX) status on the port. Hubs and switches are deliberately wired the opposite of the way end stations are wired, so that when a hub or switch is connected to an end station, a straight through Ethernet cable can be used, and the pairs are matched up properly. When two hubs or switches are connected to each other, or two end stations are connected to each other, a crossover cable is used to ensure that the correct pairs are connected. The possible field values are: MDIX

Use for hubs and switches. MDI Use for end stations.