010844 V1 VoIP IP Speaker (Replacement Product is 011394)

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.

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 CyberData Support 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 specified in our Support Knowledgebase article, Connecting to Compatible IP-PBX Servers.

(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.

Please see our CyberData Technical Support Knowledgebase article, Asterisk Settings for Speakers.

See the CyberData Technical Support Knowledgebase article, Setting up a Page Group in Asterisk.

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.

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: MAST can handle ulaw and alaw as well as streaming from a mic to a speaker. 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)

G711 u-Law or A-law, 8 bit, 8000 hz, 160 byte packets every 20ms.

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.

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.

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.