search for: ht286

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, but hear nothing (just dead ear). 2. When HT use G729 codec => it gets busy signal and I could see such output on aste...

I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays "incoming call" on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried diff...

I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes

...og GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how do i do this in Asterisk? Basically Asterisk should dial the extension number and then send required number as DTMF tones to the Gateway through the ATA. I am using FreePBX, which allows me to create a custom trunk f...

...ime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf: [ht286] type=friend regexten=6010 username=ht286 secret=secret context=numberplan-local callerid="Home Phone" <6010> host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw allow=gsm mailbox=6010 at default dtmfmode=rfc2833 extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG...

...e building, can I use some T.38 ATA's and Asterisk 1.4 to hook up my fax machines and save me the hassle of running copper all over just for fax machines? Or, am I better off just running the copper. I've done a few hours research and here's what I was thinking: FAX A <-> GS-HT286 <-> |----------| | | |--------| <-> POTS A FAX B <-> GS-HT286 <-> | Asterisk | | GS | <-> POTS B | 1.4.2 | <-> |GXW-4104| <-> POTS C FAX C <-> GS-HT286 <-> | to route |...

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:017070895...

I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodical...

...which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use the reinvite i don't have the T38 fax support. As i understood asterisk lack the T38 support for the moment. As you know there is any T38 fax support in Grandstream HT286/486? Thanks for you help, Bye, Marcello

Hello all, I'm a newbie in * and i want to start by making internall calls between ip phones (Grandstream BT100, and HT286), if someone can help me with an ewample of sip.conf file specially with the "register" field in [general] defintion. Thanks D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail sur http://fr.mail.yahoo.com/

Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/aster...

Hello, fax/ata(ht286) -> asterisk/tdm04b -> pstn fax machine I can fax out from the sip side, but I can't fax in from the PSTN side. When I try to send a fax, asterisk sees the call and show me this: "Redirecting Zap/1-1 to fax extension" "Timeout on Zap/1-1" TCPDUMP doesn't show a...

...oblem when getting fax call, which call will become a normal phone call and no fax was printed. When fax is detected, the fax extension is executed and dial the extension of the HT486 device (firmware 1.0.5.22). Somehow sending fax out working well. In the mailing lists, i notice some are using HT286 and it work. Could someone share theirs experience and give some help? E1 PSTN ----------> Asterisk (TE100P) --------> HT486 ----> Fax machine zapata.conf [channel] context=Local switchtype=euroisdn signalling=pri_cpe rxwink=300 callwaiting=yes usecallingpres=yes; callwa...

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

...o make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten => s,1,Macro(stdexten,106,SIP/ht286,t) [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten => s,1,Answer exten => s,1,Background(enter-ext-of-person) exten => s,n,WaitExten(20) exten => 100,1,Dial(Zap/1,30) exten => 106,1,Macro(stdexten,106,SIP/ht286) exten => 101,1...

Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug

Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler

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've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack