Equipment – VoIP
Download and extract the firmware file for your VoIP product type.
Then follow the instructions in the related Setup & Configuration Manual to your product type.
The first step is to fill out and submit the linked VoIP Help Desk Request Form which will help to define the issue and gather all pertinent information needed to expedite the process.
Contact Guardian Telecom if one-on-one technical help is required.
When dialing the three-digit DTMF tone on the IP phone, I can hear the DTMF-tones coming out of the speaker of the VoIP device but there is no relay action. The relay works when using the relay test-button on the configuration software. How do I fix this?
Since the relay test button is working, it seems like the problem results from interfacing with the IP phone where the DTMF tone is generated. To resolve this problem verify that the DTMF tone on the phone is set to out-of-band.
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).
This is a common problem when the re-registration time value is not set correctly.
On a Guardian VoIP 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.
On an Asterisk-based VoIP SIP PBX system, the Guardian SIP Device status is “Busy” or “Unreachable”. I have set up both the Guardian 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 Guardian 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.
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
You can use the free utility ‘Audacity’ (http://audacity.sourceforge.net/) to convert audio files into a format the device can recognize.
When you export an audio file with this program, you can save the output as “WAV (Microsoft) signed 16 bit PCM.”
Guardian Telecom VoIP units feature a built-in “fail safe” mechanism.
The Device will store the “TFTP Server IP” and “New Filename” entered on the “Update Firmware” web page. If, during the boot process, the Device is unable to boot the firmware, it will attempt to download the stored image from the stored TFTP server.
I see in the electrical connection diagram in the user’s guide that there is a High PIV Ultra Fast switching diode. Do I need it and if so do you have a source?
This High PIV Ultra Fast switching diode prevents CEMF kick back from an intermediary relay coil when power is cut and the coil field collapses. You could use an On-Semi MUR105 diode or an IN4007, which is readily available.
We have the Cisco 3550 switch and it looks like the unit is not able to negotiate the power with the switch. It keeps cycling over and over.
This happens because with default settings, the switch port is resetting power too quickly. Therefore, on the 3550 switch, on the switch port that the unit is attached to, please try adding the following CLI command:
power inline delay shutdown 20 initial 300
That should keep power supplied until the unit can boot up all the way.