Discussion: Grandstream Handytone-486

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Re: Does it have a real FXO port?

i have same problem did any one find how we can make it work as real voice gateway i mean i want to transfer my call from voip to pstn ? is there any way what about 488 does do that if yes plz any one can tell me where i can find? thx plz send me email to hayik@hotmail.com

by habibhayek, Friday 20 of March, 2009 (12:36:24 UTC)
No FXO Port!

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.

by greyfox, Wednesday 24 of January, 2007 (00:56:08 UTC)
Registering Trouble - I'm in town!

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!


by m, Thursday 19 of January, 2006 (04:12:09 UTC)
Re: Does it have a real FXO port?

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.

by m, Thursday 19 of January, 2006 (03:01:10 UTC)
How do I recover password?

I forgot ADMIN mode password.
How do I salvage configuration?
Please help me.

by smith99, Wednesday 10 of August, 2005 (04:38:01 UTC)
Re: Call forwarding

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?)

by raster, Friday 20 of May, 2005 (16:20:50 UTC)
Only the ulaw and alaw codecs works

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

WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs!
Feb  3 11:17:15 NOTICE[9529]: 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.

by , Thursday 07 of June, 2012 (05:09:29 UTC)
Re: Does it have a real FXO port?

For sure its not a "hop on, hop off" FXO. just can be used for fallout cases.

Kamran Zaidi

by , Sunday 31 of October, 2004 (04:16:20 UTC)
Loss of connection

(: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.

by , Tuesday 21 of September, 2004 (21:00:07 UTC)
Re: Unregistering

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.

by , Tuesday 21 of September, 2004 (20:52:42 UTC)