Frequently Asked Questions
1. How do I update my firmware?
2. Are your speakers compliant with RFC 3261?
3. Do your speakers support other protocols?
4. Our IP-PBX server is RFC 3261 compliant in that it can register other SIP endpoints, so how do I create multiple paging zones using your speakers?
5. Which IP-PBX servers do your speakers interoperate with?
6. I hooked up your speakers using Asterisk and they play audio individually but why don’t they play in a paging group (zone)?
7. For additional support or answers to questions not covered on this page, who should I contact?
8. What are the Asterisk settings to set up our paging speakers?
9. How do I set up a page group in Asterisk?
10. Are you able to traverse the NAT with your IP paging products?
11. How do I configure Music-on-Hold (MOH) for Asterisk?
12. We would like to communicate directly to your products. Do you have any open source programs you can recommend to do this?
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 Ceiling Speaker Operation Guide.
2. Are your speakers compliant with RFC 3261?
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.
3. Do your speakers support other protocols?
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.
4. Our IP-PBX server is RFC 3261 compliant in that it can register other SIP endpoints, so how do I create multiple paging
zones using your speakers?
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.
5. Which IP-PBX servers do your speakers interoperate with?
Our speakers interoperate with the IP-PBX servers in the following link.
6. I hooked up your speakers using Asterisk and they play audio individually but why don’t they play in a paging group (zone)?
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.
7. For additional support or answers to questions not covered on this page, who should I contact?
Contact support@cyberdata.net
8. What are the Asterisk settings to set up our paging speakers?
For both ceiling speaker and the Loudspeaker 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>
9. How do I set up a page group in Asterisk?
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]
10. 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.
11. How do I configure Music-on-Hold (MOH) for Asterisk?
Use the instructions at the following link to set up MOH for Asterisk:
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
12. We would like to communicate directly to your products. Do you have any open source programs you can recommend to do this?
Yes. Try the following links and also contact CyberData support for help Data format: RTP audio 8k G.711 a/u Law 30Ms 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).
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