011411 Syn-Apps Enabled Paging Amplifier

IP multicast is a technique for one-to-many communication over a local IP network. IGMP multicast is sent via UDP packets from a multicast server to a multicast receiver. This stream scales to a larger receiver population by not requiring prior knowledge of who or how many receivers there are. Multicast uses network infrastructure efficiently by requiring the source to send a packet only once, even if it needs to be delivered to many receivers. The network replicates the packet in order to reach multiple receivers only when necessary.

Multicast is a great alternative to SIP registration as it does not require that the recipient of the stream to register as an extension on an IP-PBX server.

CyberData products support both regular IGMP multicast as well as proprietary Polycom Group Paging.

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 the Upgrading the Firmware section of the VoIP Paging Amplifier Operations Guide.

Contact CyberData VoIP Technical Support.

CyberData's VoIP Paging Amplifier supports the following outputs:

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 IP 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>

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.

What 8 ohm speakers are recommended for the Paging Amplifier?

My speakers are not loud enough. What should I do?

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:

https://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:

Linphone: https://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 (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.

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.

The speaker calculator on your website seems sufficient for square rooms. It is hallways that I am struggling with. The hallways measure 9 feet x 9 feet by 100-150 feet in length.

In a ceiling mount, you would want about 1 per every 10-15 feet of hallway length.

If you used the paging amp you could use 3 paging amps per 100 feet as per this drawing.

This assumes average ambient noise and surface attenuation levels.

The following figure shows the 011061 V2 Paging Amp and the 011068 Horn Loudspeaker:

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:

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:

I have a Cisco 6513 switch. When I plug in a CyberData VoIP device, the device constantly reboots and will not register to the SIP server.

Please adjust the switch power selection to Power inline static, as opposed to Power inline dynamic. This will allow the device to continuously receive 15.4W of power.

The V2 Paging Amplifier's Line Input and Output can be connected to a 600 Ohm Lo-Z input or output using a Hi-Z to Lo-Z impedance matching transformer. For example, the Bogen WMT1A input matching transformer is an easy to use, inexpensive solution that does not require soldering or complex wiring.

Create a second directory number and user for the Nightringer extension. You may need to fudge a digit of the MAC Address so that Call Manager allows you to associate it to the new user. Be sure this MAC address does not match any other CyberData devices on your network.

There are known PoE limitations with third party devices on the Cisco Catalyst series switches. This includes the 2500, 3500, and 6500 series switches. It would be best to avoid using the Cisco Catalyst series switches to power the amplifier and use an 802.3at injector. If you have a 2960-S series switch with PoE+ ports, we recommend that you use those instead of PoE and set the amplifier to High Power mode.

See the following webpage for a nice selection: https://www.sti-usa.com/pcategories/waterproof_buttons_and_switches/