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Ports
- 1 Power (5V)
- 1 RJ45 WAN
- 1 RJ45 LAN
- 1 RJ11 FXS
- 1 RJ11 FXO/PSTN
This is (mostly) a step up from the Grandstream Handytone-486
While unpowered, the PSTN RJ11 is directly connected to the FXS RJ11.
When powered, the analog phone is is disconnected from the PSTN line. Dialing "*00" (configurable) causes a relay to join the analog phone to the PSTN to force outgoing calls via PSTN. Plus, up to 5 dial patterns ("911") can be programmed which go out via PSTN. Otherwise dialing is done via the FXS VOIP.
When a call arrives via the FSX (VOIP) port or the PSTN port, the attached analog phone rings.
Bugs:
- The "early dial" feature does not work. E.g. if you want it to pass each dialed key to the PBX as it is pressed rather than taking your whole number (followed by the '#" key or a 4-second timeout) before sending to the PBX, it doesn't work. Well known Grandstream bug in many of their products. With Asterisk, pressing the 2nd digit with early dial enabled, causes an instant dial failure.
Issues:
- The HT-488, unbelievably, does not pass CallerID from the incoming PSTN call through to the PBX. It does pass the CallerID over to the FSX line and is visible on a CallerID-aware analog phone.
- Unlike the HT-488, the "LAN" port is only NAT'd behind the WAN port. On the HT-486, one can choose to have the LAN port switched(bridged) to the WAN port or NAT'd.
- For outgoing calls from PBX through the FXO port, the actual number dialed in the SIP INVITE is ignored. Instead the HT-488 "answers" the call itself and waits for an optional PIN followed by the destination PSTN phone number before actually taking the PSTN line off-hook and actually dial. Not a big deal for Asterisk. Use a the "D" flag to the Dial application. Don't forget to use a leading 'w' on the number to wait a bit before dialing.
- The HT-488 does not support Disconnect Supervision. It only supports silence detection and disconnect tone detection. Grandstream is working on this: "We are working actively now to get the PSTN disconnect issues solved. We do detect polority reversal in our current firmware, but it is not working properly with the released firmware."
- Incoming PSTN calls must ring the FXS port at least once before being forwarded to VoIP.
Nice features:
- Unlike the HT-486, the analog phone connected to the FXS port can access the PSTN line with more than just the "*00" pattern. It allows for things like "911". In my mind this one feature makes the HT-488 useful for applications where you would have used the HT-486. Just ignore the FXO port of the HT-488.
- The Voicemail Waiting Indicator works fine. The button on the top of the unit blinks green slowly.
- Unlike the Digium T4xxP cards, the HT-486 returns a SIP BUSY status if there is no phone line connected to the FXO port, or if there is a call in progress already.
Grandstream's page about the HandyTone-488
Grandstream Support Forum
A configuration guide for Asterisk and HT-488
Other Grandstream products
Places to buy:
- Africa (All) - VOXSYS,RSCE - Buy on line, all Grandstream products at wholesale prices (Portugal and Spain based, delivers to all Europe and Africa).
- America - 888VoipStore.com - Best Prices on Handytone 488 HT-488. Call for volume pricing. 888-VOIPSTORE.
- America - Grandstream HandyTone 488 - VoIP Supply is leading supplier of the Grandstream HT-488.
- PBXSelect Authorized Grandstream Reseller - FREE US SHIPPING ENTER "WIKIFREESHIP" AT Checkout!
- Belgium - VoIPsolutions At 54 Euro (payment in EUR, resellers prices avaible) Delivery all of Europe
- Israel- VoIPstore.co.il
- Portugal - VOXSYS,RSCE - Buy on line, all Grandstream products at wholesale prices (Portugal and Spain based, delivers to all Europe and Africa).
- Spain - VOXSYS,RSCE - Buy on line, all Grandstream products at wholesale prices (Portugal and Spain based, delivers to all Europe and Africa).
- USA - discountvoipoutlet.com
- Worldwide - The VoIP Connection
- Worldwide - Voxilla Store - Grandstream HandyTone ATA-488
- Ukraine - Wildix in Ukraine
Go back to VOIP Gateways
Page Changes
won't reboot
FXO -> FXS Ring-through suppresion; no Caller ID CID passthrough, ever
Also, I sent an e-mail to Grandstream support regarding the Caller ID passthrough issue (and the ring-through issue, among other things). Their response to the Caller ID question:
"Unfortunately, as result from luck of resourses the CID information can not be transfered from the PSTN line to VoIP network. This feature will be available in our new generation HT-503 device." (not edited for spelling or grammar).
So, for my purposes, this device doesn't quite meet my needs (Caller ID CID info passthrough is the deal breaker). So, I've just ordered a Cisco/Linksys SPA3102, and probably going to use this device as an FXS somewhere else. :| The Budge Tone 200 phone I bought is pretty satisfactory, however.
Direct dial problem
The Caller ID forwarding problem doesn't appear to be fixed, however, and it is annoying.
One channel at a time
I assumed that the FXO port on the HT-488 would gab incoming calls on the 1st ring and forward them by SIP to Asterisk, and this works well. I have trained everyone not to answer on the 1st ring. I also assumed that I could call the separate SIP FXS account on the HT 488 from Asterisk and make all the phones in my house ring. This works too.
However, what makes this all worthless is that the HT-488 can only do one of these things at a time. When a home incoming call from the telco is forwarded by the HT-488 to Asterisk, I try to return the call back to the HT-488 FXS port to make the home phones ring. When I try to start a SIP connection to the FXS port I get a busy indication from the HT-488. This is a feature, I suppose. You've been warned.
So, while the HT-488 has 2 SIP accounts and registers as 2 peers, it can only use one of these at a time.
New firmware destroyed unit
Update: The Grandstream tech support guy emailed me back. We have made a good bit of progress, but the unit is still non-functional. It thinks its registered, but its not. It was necessary to load version 77 of the firmware. The LED no longer blinks funny colors, but because its not registering with the asterisk box, its not working.
Product No Good
So if you're considering this product, stop and consider the SPA3102 instead. The extra $30 will more than pay for itself.
This device is not suitable for production
Upgraded 2 of them to newest 1.0.3.18 firmware. And put all 4 into company which were making a lot of calls.
Devices were used about 80-90% of the time (work hours). So almost all the time they were under heavy load.
Then it started. They began to die after different amount of time. Some of them kept working for all day, then die next day, some after 2-3 hours.
They just stop responding. Does not answer to ping. Nothing. Plain dead. Only restart could help them.
We played with them for 2 weeks! (Yes, we have a nice client and some backup solution for him). Power supply is ok, not a special PSTN line, nothing unusual. Searched Google till death, tried every possible combination of settings - nothing helped. 4 devices kept dying and dying and dying.
I believe it's a nice toy for persons who are receiving/making 5-10 calls per day, but for serious production it's totally unusable.
I strongly not recomend using these devices in production environment.
BTW: CallerID PSTN->VoIP is NOT working
PSTN pass through
Also when in PSTN pass through mode are the ports bridge via the interal relay or is the connection internally FXO -> VoIP -> FXS? I'm looking for a device that will enter PSTN pass through by physically bridging the ports so echo is not the resposibiity of the ATA.
Caller ID is working
CallerID *may* come in the future