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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
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Grandstream Handytone-486

HandyTone-486


Image

Ports

  • 1 Power (5V)
  • 1 RJ45 WAN
  • 1 RJ45 LAN
  • 1 RJ11 FXS
  • 1 RJ11 PSTN (Not a true FXO -not usable via SIP, only used as a fallback for power failure, etc)


Basically this is the big brother of Grandstreams Handytone-286. The big difference is that you can hook your regular phone line in there and that it can act as a NAT Router in your home.

The LAN port can either be switched or NAT'd behind the WAN port. (Configurable. Very cool. The HT-488 only does NAT.)

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. 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. CallerID is passed through from the PSTN line and, if provided on the VOIP call, is generated out down the analog line to your CallerID-capable analog phone.

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 all of their products. With Asterisk, pressing the 2nd digit with early dial enabled, causes an instant dial failure.

Major oversight:
  • Unlike the HT-488, the only way for the analog phone to access the PSTN line on the HT-486 is via the "*00" prefix. The HT-488 allows up to 4 other programmable numbers (like '911', duh) to automatically use the PSTN line.


Grandstream's page about the HandyTone-486

Grandstream HandyTone-486 Review

Grandstream Support Forum



Available from:


Configuration Geometries:

LAN inhabitant with NAT router:

 phone       Internet
 line           |
   |          (WAN)
   |       NAT Router
   |          (LAN)
   |            |
 (line)  (wan)--+-----other devices
 Grandstream
 (phone) (lan)
   |
 handset

LAN using the Grandstream as a NAT router:

 phone  Internet
 line      |
   |     (WAN)
   |     Gateway
   |     (LAN)
   |       |
 (line)  (wan)
 Grandstream
 (phone) (lan)
   |       |
 handset   other devices





Grandstream Handytone-486
Other Grandstream products

Go back to VOIP Gateways

Created by Martin List-Petersen, Last modification by ANTHONY BANDINELLI on Mon 03 of Mar, 2008 [17:08 UTC]

Comments Filter

No FXO Port!

by Bruce Ide on Wednesday 24 of January, 2007 [00:56:08 UTC]
If you're looking to add, say, an FXO port to an Apple Mac keep looking. This thing DOES have a line-in port but it's not actually an FXO port. It's simply a ring-through to an analog telephone. The documentation on this thing is terrible and configuration is not much fun. You can get it to register with asterisk if you're looking for a $50 FXS port but otherwise I'd suggest giving this thing a miss.

Re: How do I recover password?

by imtiaz ahmad on Friday 09 of June, 2006 [02:55:08 UTC]

Registering Trouble - I'm in town!

by m on Thursday 19 of January, 2006 [04:12:09 UTC]
I'm having trouble registering my Grandstream 486. Can anyone help please?

My setup:

I have _nothing_ hanging off the grandstream LAN port. The WAN port is attached to my teency local network, with addresses of the form x.y.z.*. Also on the local network are some computers and an ADSL router/gateway. The router runs NAT and its address on the local network is x.y.z.55.

In order to keep life simple by keeping down the number of address spaces I've set the grandstream into bridge mode. I've set the grandstream LAN network mask to 255.255.255.255 and address to x.y.z.206, an otherwise unused address. That should stop the LAN port from playing any part in the proceedings!

I've set the grandstream WAN port address statically to x.y.z.99 and the network mask, DNS server and router to 255.255.255.0, x.y.z.55 and x.y.z.55 respectively.

What I can do:
- I've set up the X-lite softfone on one of the computers on the local network and I can ring from it to the budgetone, no problem. I can't ring back.

What I can't do and I'd really like to do is to register. With Nikotel. I've set up my user ID, password and STUN settings as per Grandstream's recommendation for Nikotel but no joy.

Questions:

Is STUN needed when making outgoing calls? Surely it's just a way of making incoming calls possible? If so, is there a way that I can test that I can do direct IP calls to outside my local network? That would be a start.

My current ADSL router runs NAT but doesn't seem to support manually configuring the port forwarding. One thing I could do is to chuck it and replace it with one that does. Presumably if I could point the SIP ports at the grandstream I wouldn't have to mess around with STUN, right? However it would still be nice to get the grandstream working with what I've got now!

Re: Does it have a real FXO port?

by m on Thursday 19 of January, 2006 [03:01:10 UTC]
Well, the FXO port is real enough in that it will accept an inbound call, but it always routes it to the phone. I believe that you can't pipe it out to either the WAN or the LAN port. At least, if it's possible I can't see how.

Regards, M.

How do I recover password?

by smith99 on Wednesday 10 of August, 2005 [04:38:01 UTC]
I forgot ADMIN mode password.
How do I salvage configuration?
Please help me.

Re: Call forwarding

by raster on Friday 20 of May, 2005 [16:20:50 UTC]
Same problem here... Call forwarding does not seem to work. (It should forward to *any* PSTN or mobile number right? It does not assume a SIP number, does it?)
Edit

Only the ulaw and alaw codecs works

by Anonymous on Thursday 03 of February, 2005 [11:00:09 UTC]
The grandstream ATA 486 schould support almost all codecs,
but it doesn't work in my case. The ulaw and alaw takes with
overhead almost 72kb traffic so it is not possilbe to use it on 256/64 internet
connection for example.
I get the following message when I force the use of different codec

WARNING9529: chan_sip.c:2765 process_sdp: No compatible codecs!
Feb 3 11:17:15 NOTICE9529: chan_sip.c:7395 handle_request: Unable to create/find channel

What could I do to see some more detailed logs?

My sip.conf

p1
type=friend
username=p1
fromuser=p1
dtmfmode=rfc2833;info;inband;info;rfc2833 ;inband info http://www.voip-info.org/wiki-
Asterisk
secret=
host=dynamic
amaflags=default ; Choices are default, omit, billing, documentation
allow=all

Has anybody experienced this?

I was trying to change almost anything with the some result.

In the granstream configuration webpage are the following
things to configure, I don't understend, maybe it could do that tric.
G723 rate: 6.3kbps encoding rate 5.3kbps encoding rate // tried both
iLBC frame size: 20ms 30ms // 20 ms
iLBC payload type: (between 96 and 127, default is 98)//98
Silence Suppression: No Yes
Voice Frames per TX: 2 (up to 10/20/32/64 for G711/G726/G723/other codecs respectively) // did not try
Layer 3 QoS: (Diff-Serv or Precedence value) // 48
Layer 2 QoS: 802.1Q/VLAN Tag 802.1p priority value (0-7) /0 0

I'm using the 1.0.5.18 firmaware, and was using the buggy 1.0.5.21

My asterisk is working fine with about 8 SIP and IAX2 providers using any codecs ...
(also the 723) I'm using about 1 month old Asterisk from the CVS.

Any comments would be appreciated.
Edit

Re: Does it have a real FXO port?

by Anonymous on Sunday 31 of October, 2004 [04:16:20 UTC]
For sure its not a "hop on, hop off" FXO. just can be used for fallout cases.

Kamran Zaidi
Edit

Loss of connection

by Anonymous on Tuesday 21 of September, 2004 [21:00:07 UTC]
(:cry:) I have been trying all kinds of combinations of settings on my ATA for three weeks now.

How do I get my ATA to continually try and reestablish service with my SIP service provider? I know that NTLWORLD (my ISP) bring down various parts of their network (including DHCP and DNS) servers for several minutes at a time two to three times per day. This drives routers and my ATA-486 bannanas! Only rebooting the device from the web-interface or yanking the power cable out for a few seconds restores the service.

I have the following configuration:

Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6

WAN IP Address: dynamically assigned via DHCP
Use this DNS server (if specified): . . . (have tried specifying this but lose service when this DNS goes down)
SIP Registration: Yes
Use DNS SRV: No (have tried Yes)
Unregister On Reboot: Yes (Have tried both Yes and No here)

I do not have a static IP and I don't have a PPPoE account!

Any help regarding network connection problems would be appreciated.

Thanks

Dave B.
Edit

Re: Unregistering

by Anonymous on Tuesday 21 of September, 2004 [20:52:42 UTC]
The product is pretty good but the documentation sucks.

Firstly check the firmware version you are using (the latest is Software Version: Program--1.0.5.11 Bootloader--1.0.0.18 HTML--1.0.0.37 VOC--1.0.0.6 ) search on GOOGLE for "Grandstream 486 firmware". Secondly try selecting "Yes" for 'unregister on reboot'.

Does anyone know what the use DNS option really does? At the end of the day the device has to translate the names used for the SIP service some how!(:confused:).

I am having problems with loss of service. I can't work out whether it is a DNS problem (NTLWORLD my ISP keep bringing down their DNS servers for extended periods of time) - there are also periods when the resolving of DNS names takes a PC over 20 seconds. When DNS is there and name translation is quick there is no problem.

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