similar to: Grandstream HT-286 review on the way

Hi List, Just received a HT-286 Analog Telephone Adapter. This device is allows the user to take a standard analog telephone set and connect it to a VoIP/SIP based gateway. A low cost way of having PSTN devices make use of VoIP/SIP services. For example you could take a credit card processing machine and punch it into a location where you have broad band access. You could take a FAX

I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the "DMZ Host" in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 ====== NAT FW ====== * Server

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Hi, I have a grandstream HT 503 ATA which seems to support TLS/SRTP. I generated the keys and certificates for both the server and the device as mentioned at https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial I then copied the grandstream.crt and grandstream.key over to the ATA. Unfortunately, the device is refusing to register and in asterisk cli i see SSL certificate ok ==

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

Hi, I've had a bunch of people asking about purchasing GS phones. I'm happy to announce that we now have an online purchase system available. We have: BT-101 BT-102 HT-286 Plus the full line of Digium hardware. We will be adding other products in the near future so check back :) Buyers are responsible for s/h costs. The URL for the phones is:

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

List Mom/Dad, could you please find in the list of subscribers the person that is getting mail at uol.com.br or something similar and smack them for having broken Anti-Spam software ?? I'd be happy to see them just be removed until they fix it. cheers ----- Forwarded message from AntiSpam UOL <andersoncbr.sspam@uol.com.br> ----- Delivered-To: jmbrown@chagresventures.com Date:

Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things

Hello there, I just wasted some time setting up a Grandstream HT-488 to be used with Asterisk, so I thought I'd share the experience by writing a small tutorial at: http://astrecipes.net/index.php?n=338 Most tutorials I came across were for old versions of the firmware, and I spent too much time trying to figure out what the Forward to VoIP option wanted.... :-( I hope this helps,

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

I have an analog Fax machine which I wish to connect to the network and the Asterisk server. It will connect through a GS Handytone 286 converter and then into the LAN. Is there any information out there on what I need to write in *sip.conf* and/or *extensions.conf* to make sure the fax works as a fax? Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do I need to

I'ld like to test SMSes between Asterisk and an analog phone on a GS-ATA-286 (SIP). I have spent many hours trying any exemple on voip-info or mailing-list, but no message got sent from the analog phone. My goal, at now, is to send SMSes locally from and to Asterisk. Does anyone have a working configuration ?

Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme