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For questions not answered here, please use our Contact VoIP Support Form to contact us.
1. How do I update my firmware?
2. For additional support or answers to questions not covered on this page, who should I contact?
3. What does this product do?
4. Are you able to traverse the NAT with your IP paging products?
5. I was able to register your device with our SIP server, but when I tried to enter a DTMF tone there was no function.
6. After a period of time, my device stops working or is unreachable.
7. On an Asterisk-based VoIP SIP PBX system, the CyberData SIP Device status is "Busy" or "Unreachable".
8. I have set up the CyberData Paging Server per the instructions along with the CyberData VoIP speakers. I can ping the Paging Server, pull up the server web pages, and see that it is registering with the SIP server. However, when I dial the SIP extension, the Paging Server answers, but then appears to hang. There is no ready tone and no paging happens when I dial the mgroup zone paging DTMF tones.
9. I upgraded my 3CX PBX server to 7.1 and now my Rev B CyberData VoIP Ceiling speaker and My VoIP Paging Amplifier do not stay registered with the server.
10. If I wanted to power the device by using the auxiliary power connector, what type of power supply do I need?
11. I am unable to connect with the unit when it is plugged into a Cisco SLM 224P switch.
1. How do I update my firmware?
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 speaker firmware from your PC, see Section 2.4 of the VoIP Paging Server Operation Guide.
2. What does this product do?
Our Paging Server is a POE enabled, single SIP-endpoint enabling user defined paging zones through a multicasting connection to CyberData VoIP speakers.
3. So how does this product support 99 zones?
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.
4. Are you able to traverse the NAT with your IP paging products?
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. For more information on SBCs, go here.
5. I was able to register your device with our SIP server, but when I tried to enter a DTMF tone there was no function.
Make sure your SIP phone is set to 101 for the DTMF payload type (Out of Band RFC2833).
6. After a period of time, my device stops working or is unreachable.
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.
7. On an Asterisk-based VoIP SIP PBX system, the CyberData SIP Device status is "Busy" or "Unreachable". I have set up
both the CyberData VoIP SIP device and the PBX extension information for the device. I can see the device on the network,
am able to PING it, and can bring up the device web page with a browser. However, when I try to call it from a phone extension,
I see the word "Busy" or "Unreachable" in the Asterisk log.
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 (3 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.
8. I have set up the CyberData Paging Server per the instructions along with the CyberData VoIP speakers. I can ping the Paging Server, pull up the server web pages, and see that it is registering with the SIP server. However, when I dial the SIP extension, the Paging Server answers, but then appears to hang. There is no ready tone and no paging happens when I dial the mgroup zone paging DTMF tones.
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 by completing the following steps:
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.
9. I upgraded my 3CX PBX server to 7.1 and now my Rev B CyberData VoIP Ceiling speaker and My VoIP Paging Amplifier do not stay registered with the server.
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.
Note: There are pictures of the 3CX PBX pages and parameters at the end of the document at this link.
10. If I wanted to power the device by using the auxiliary power connector, what type of power supply do I need?
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.
11. I am unable to connect with the unit when it is plugged into a Cisco SLM 224P switch.
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.
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