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 Amplifier Operation Guide.

Contact CyberData VoIP Technical Support.

CyberData's VoIP Paging Amplifier supports the following outputs:

One 8-ohm speaker: 8 watts
One 8-ohm speaker with external PoE power injector (CyberData PN 010867): 10 watts
Two 8-ohm speaker in parallel with external PoE power injector (CyberData PN 010867): 16 watts
The Paging Amplifier does not support three loudspeakers in parallel, since this would result in less than 4 ohms impedance and overdrive the output driver.

Yes, our speakers are compliant with RFC 3261, but not every SIP extension is fully supported such as extensions for certain phone features that our speakers do not require.

Our speakers are SIP endpoints that use the SIP protocol in RFC 3261. Depending on the business case, we will consider custom applications using other protocols. Please contact sales@cyberdata.net for inquiries concerning other protocols.

Our speakers do not create multiple zones as this is a feature (SIP extension) of the IP-PBX server. If your IP-PBX server does not support this SIP extension, you can use our Paging Server product to create multiple zones with our speaker.

Our speakers interoperate with the IP-PBX servers in the following link.

1)Make sure you have installed and loaded a timing source such as Zaptel’s “ztdummy” on your Asterisk server.

2) If you are using SIP phones in the same paging group as our speakers and auto-answer is activated for these phones, please upgrade to the latest paging group module in Asterisk, which is 1.2.3 or greater and put an “x” (this removes auto-answer commands our speakers do not use because they are hard-coded to auto-answer) after the extension number for the speakers in the paging group drop-down menu in FreePBX.

For both ceiling speaker and the Paging Amplifier, the settings are contained in the sip_additional.conf file in Asterisk. This file should read as follows in which "100" represents the username of our speaker and the password (secret) is ext100:

[100]
username=100
type=friend
secret=ext100
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=100@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=100 <100>

A: Page groups are found in the extensions_additional.conf file in Asterisk. The paging group portion of this file should read as follows in which "101 and 102" are the usernames and extensions of two speakers and "103" is the extension used to call the paging group of speakers 101 and 102, and "105" is the extension to call speakers 101,102, and 104:

[ext-paging]
include => ext-paging-custom
exten => PAGE102,1,GotoIf($[ ${CALLERID(number)} = 102 ]?skipself)
exten => PAGE102,n,Dial(SIP/102,5)
exten => PAGE102,n(skipself),Noop(Not paging originator)
exten => PAGE101,1,GotoIf($[ ${CALLERID(number)} = 101 ]?skipself)
exten => PAGE101,n,Dial(SIP/101,5)
exten => PAGE101,n(skipself),Noop(Not paging originator)
exten => PAGE104,1,GotoIf($[ ${CALLERID(number)} = 104 ]?skipself)
exten => PAGE104,n,Dial(SIP/104,5)
exten => PAGE104,n(skipself),Noop(Not paging originator)
exten => Debug,1,Noop(dialstr is LOCAL/PAGE102@ext-paging&LOCAL/PAGE101@ext-paging)
exten => Debug,n,Noop(dialstr is LOCAL/PAGE104@ext-paging&LOCAL/PAGE102@ext-paging&LOCAL/PAGE101@ext-paging)
exten => 103,1,Page(LOCAL/PAGE102@ext-paging&LOCAL/PAGE101@ext-paging)
exten => 105,1,Page(LOCAL/PAGE104@ext-paging&LOCAL/PAGE102@ext-paging&LOCAL/PAGE101@ext-paging)

; end of [ext-paging]

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.

Since the VoIP Paging Amplifier is a single SIP endpoint, it can not have multiple paging groups.

Yes, our loudspeaker outputs are potential free. If no paging input occurs, the outputs is at or near zero voltage.

The output level is 15 Volt maximum.

• CyberData's 011068 Loudspeaker.

CyberData recommends good quality speaker wire that is 12 guage and a maximum length of 25 feet.

For maximum output, use two 8-Ohm speakers that are wired in parallel with the shortest possible cable length (total combined length should not exceed more than 25 feet).

By utilizing the high power jumper settings described in the manual in conjuction with CyberData's PoE Injector, you may achieve additional speaker volume by using either one 8-Ohm speaker or two 8-Ohm speakers in parallel.

Use the instructions at the following link to set up MOH for Asterisk:

http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf

Yes. Try the following links and also contact CyberData support for help Data format: RTP audio 8k G.711 a/u Law 20ms packet time Open source programs:

MAST: http://www.aelius.com/njh/mast/
MAST can handle ulaw and alaw as well as streaming from a mic to a speaker.

Linphone: http://www.linphone.org/index.php/eng
Linphone has a test application called rtpsend that streams data out as ulaw by default. (It may be trivial to modify this to send in other formats).

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 (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.

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.

The connection of two 8 ohm speakers in parallel under both Standard and High Power Mode require the use of the CyberData PoE injector:
http://www.cyberdata.net/products/voip/accessories/poeinjector/index.html