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HT488 FAQs » How is the HT488 different from other HT models? » How is the HT488 used? What is the typical configuration? » How do I make or receive PSTN calls using only the PSTN line? » Can you explain the use of ‘PSTN pass through’ and ‘FXO port’? » What is the parameter “Number of Rings”? » What is the parameter “Forward to PSTN”? » What is the parameter “Route call to PSTN”? » How do I configure the HT488 to send and receive calls between the US and another country using a PSTN line and VoIP? » How do I configure the HT488 to send and receive calls between the US and another country using a PSTN extension and a VoIP desk phone using the Internet? » How do I configure the HT488 to recognize local PSTN CO disconnect signal?
How is the HT488 different from other HT models? The HT488 device is a simple gateway which enables an interconnection between VoIP and PSTN networks. The HT488 works as follows:
First, set the HT488 to the following configuration:
For VOIP to PSTN:
For PSTN to VOIP:
How is the HT488 used? What is the typical configuration? The HT488 is most often used as a ‘hop-on, hop-off’ ATA. The following example illustrates the most popular configuration for the HT488:
Assume you live in Europe and have a business in USA. Install one HT488 at your business in the US. Through the Internet, you can call this HT488 from any point in Europe, receive dial tone from the local US PSTN provider, and make US local phone calls from Europe.
Your call has been initiated in Europe, but you pay for the call as though you are calling from the US (which is significantly less expensive). The reverse is also true. You can be reached in Europe by dialing a local US number. The call will be transferred to you in Europe over the Internet.
How do I make or receive PSTN calls using only the PSTN line? When receiving a call, the phone connected to the HT488 will simply ring, provided the phone line from the wall is connected to the line port on the HT488.
To make a call, dial the ‘PSTN Line Access Code’ first, as configured on the Web Configuration Page (by default it is *00), wait for a dial-tone, and then dial the desired PSTN number.
Can you explain the use of ‘PSTN pass through’ and ‘FXO port’? PSTN Pass through port: What it can do:
What it CANNOT do:
FXO port:
What is the parameter “Number of Rings”? The parameter “Number of Rings” can be found under “Basic Settings” configuration page. The Default value of this parameter is 4. In case incoming call will arrive from the PSTN network, the analog phone connected to the FXS port will ring 4 times (default configuration). After 4 rings the HT488 will give an additional dial tone and caller can dial known VoIP extension number. This new dial tone says that original call was transferred to VoIP network and a new extension number required to be dialed. Min value of this parameter is 1. If 4 rings is not enough for you to answer the call, you can increase this value to any desired.
If you have parameter “Forward to VoIP” (at the same configuration page) configured with some VoIP extension number, call arrived from the PSTN will be unconditionally Forwarded to this preconfigured number after 4 rings.
What is the parameter “Forward to PSTN”? The parameter “Forward to PSTN” can be found under “Basic Settings” configuration page. If you have parameter “Forward to PSTN” configured with some PSTN number, call initiated from any VoIP extension to FXO extension will be automatically transferred to this preconfigured number.
What is the parameter “Route call to PSTN”? The parameter “Route call to PSTN” can be found under “Basic Settings” configuration page. If user using analog phone connected to the FXS port will dial digits configured in these fields, This call will be directly rerouted to the PSTN network. This feature is useful for emergency calls such as 911 and etc. In this field, you can configure either full number or prefix only.
How do I configure the HT488 to send and receive calls between the US and another country using a PSTN line and VoIP? EXAMPLE: I have one HT488 in the US (location X) and another HT488 in Asia (location y). I need to set them up so that I can have an incoming fixed line (PSTN) call to X forwarded to Y using VOIP and then terminate on a fixed line in Asia. How do I configure the HT488?
Fixed Line number associated with X: 444444
Fixed Line number associated with Y: 888888
Required Call flow: Incoming call in US −> 444444 −>100 −>200 −>888888 −>Fixed Line Termination in Asia
Configuration for X in US: FXS Account:
FXO Account:
(i.e. SIP Server (and/or Outbound Proxy), User ID, Auth. Password)
(i.e. PSTN AC Termination, PSTN Disconnect Tone, PSTN Disconnect Tone Cadence)
Basic Settings:
Configuration for Y in Asia: FXS Account:
FXO Account:
(i.e. SIP Server (and/or Outbound Proxy), User ID, Auth. Password)
(i.e. PSTN AC Termination, PSTN Disconnect Tone, PSTN Disconnect Tone Cadence)
Basic Settings:
How do I configure the HT488 to send and receive calls between the US and another country using a PSTN extension and a VoIP desk phone using the Internet? EXAMPLE: I need to use the HT488 as a PSTN extension over the Internet. I have an HT488 in the US and a VOIP desk phone in Asia. I want to hear the US PSTN dial tone on the desk phone in Asia, as well as be able to receive all incoming calls for US PSTN number on my desk phone. How do I configure the HT488?
Setup: You will need a VOIP endpoint (A) with “Off-hook Auto Dial feature” (example may be BT-200) and a HT488 having a public IP (or private IP but both units reside within the same LAN).
Call Flow:- A −> 5555 −> PSTN dial tone on HT488
Configuration for ‘A’:
Configuration for the HT488:
When you pick up the receiver at A, you will have a dial tone from the PSTN/PBX line through HT488. When you have a PSTN/PBX incoming call to the HT488, A will ring.
How do I configure the HT488 to recognize local PSTN CO disconnect signal? The solution depends on which disconnect type each provider implements for different countries.
In the USA, the most popular disconnect is “Current disconnect”. Using this method, the PABX initiates a short voltage drop on the line to close an existing call. This voltage drop has a specific duration. The most common duration used is 200 ms break to indicate a closed call. However, the line may experience random voltage drops. This is then interpreted by the HT488 as an end of conversation and the HT488 closes a call.
The parameter “Current Disconnect” found in the configuration options under the FXO Configuration Page corrects this problem. We recommend choosing YES for this parameter. The next parameter in the same configuration page called: “If current disconnect enabled, use threshold: ” should be configured to the optimal ms level for your specific network environment (for example, set this value to 200, 250, 300 etc). For example: by choosing 250 ms, the HT488 will ignore any random voltage drop on the line if its duration is less then 250 ms. Verify this value with your service provider or test it at different levels.
In European countries, the most popular disconnect is the “tone disconnect” method. Using this method, PSTN PABX sends a unique tone to notify the endpoint about call disconnect. The HT488 recognizes this tone using the special configuration parameter called “Enable PSTN Disconnect Tone: ”.
If this parameter is configured to “Yes”, the user should verify with the national service provider what kind of tone the PABX will send to indicate the call disconnect. Verify the frequency and cadence tones are the same using the parameters “PSTN Disconnect Tone: ” and “PSTN Disconnect Tone Cadence: ”. Once verified, the HT488 will drop the active call only in case preconfigured signal has arrived. All these configuration options also appear under FXO PORT configuration page.
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