similar to: Semi-OT: Use T.38 ATAs to Extend fax lines

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

I have an analog 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

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:0170708959@192.168.1.4;user=phone>

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime 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:

Hi there, I'm running R on windows 7 with Rstudio. Everyday I receive a zip file where a bunch of half-hourly files are zipped together. I then use xx=unzip(ind) to get xx, which consists of : [1] "./2011/A20112961503.flx" "./2011/A20112961503.log" "./2011/A20113211730.slt" "./2011/A20113211800.slt" "./2011/A20113211830.slt"

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

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,

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian

Here at the University of Sydney I''m trying to give Puppet a go. I haven''t made it past first base yet unfortunately. We run RHEL 4 here, for the most part, and the install guide at http://reductivelabs.com/trac/puppet/wiki/InstallationGuide says that you should use the vendor''s Ruby, but Red Hat''s current version of Ruby is 1.8.1-07 - is this new

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in 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

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

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 any activity to the extension that I configured to be the fax machine.

hi folks well, tinc is a really nice tool and we implemented it on 3 linux servers and 2 mobile clients (XP notebooks) so far. one of the 3 tinc servers is making troubles, when a connection is initiated from this server over a zyxel 642 adsl router out to the other 2 servers in the internet. the logfiles of the other 2 servers shows: > tinc[1398]: Received UDP packet from unknown

A DID number was dedicated to receive fax, but i have the problem 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

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,

Hi All, Need some information on getting the right storage array to buy for my cluster configuration can anyone help me with this my Setup so far 2 Athlon 2800 Sempron System 1 Wti Network power switch 1 24 Port 10/100 Cat5 switch 2 Adaptec 39160 scsi host adapter pci cards installed in the systems. I need to know what is the best storage solution with this configuration Thanks

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

Hello all, this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe table top or handheld 802.11 sip enabled phones. But the disadvantage is that how do i power each radio

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave