similar to: Grandstream HT-286 and NAT

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

Just to tease people I'll be post a URL to my review of the HT-286 by thursday..

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,

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

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

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 ==

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:

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, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Hi, I have HT Feature available on my Dell Server R720. Please help me understand if HT is enabled on the server, will there be a performance impact. What are the deciding factors to enable HT on the servers. Regards Kaushal

Dear R-help, Could someone please try to explain this paradox to me? What is more likely to show up first in a string of coin tosses, "Heads then Tails", or "Heads then Heads"? ##generate 2500 strings of random coin flips ht <- replicate(2500, paste(sample(c("H", "T"), 100, replace = TRUE), collapse =

I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps

Hi, Do we need HT enabled on CentOS release 5.5 (Final) and Kernel Version :- 2.6.18-194.el5 Please help me understand the pros and cons of having HT enabled or disabled on the Server Dell R 710. cat /proc/cpuinfo -> http://fpaste.org/K2dT/ Do let me know if anyone needs more information. Regards, Kaushal

I would like to build a package for the HT HG-U133+ PM arrays from affy, but I can't find any good documentation on how to go about it. Naively using makecdfenv's make.cdf.package() causes R to seg-fault. I'm unfamiliar with the CDF format as such, but I'm guessing that it's changed somewhat because the PM arrays no longer have P/A and mismatches. I'm looking to build

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

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

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

Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS devide