similar to: WellTech 380x Gateway

Hey all, It such a shame that BRI technology is such a flop in the USA. For a small office such as mine it would be a great product. So her goes my question.... What is a known asterisk working BRI card that will operate in the USA. I need to weigh price/quality. I need to do DID/DDI (or what ever you want to call it). Asterisk will do everything else I need. The ILEC has at the other

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

Hello all, I'm looking for software for my asterisk logs that will compile the information into nice web-based charts and graphs. Something that works similar to webalizer for apache. I want to be able to spot trends of usage, call volume levels, disconnect/failure levels, and basically see exactly where my system has been at over the past day/week/month, etc. . I would prefer for

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

Hi all I'm fairly new to Asterisk, so be nice :-) I was wondering if anyone has been able to get the Welltech K1000A USB phone working on Linux. I see audio and HID drivers loaded when it is plugged in to my Fedora Core 1 laptop, but that's about all that happens. I've searched all the usual places (FAQ, Google, etc) but not found anything helpful. Asterisk is working fine with

Recently, there're some posts about registration problems with this gateway. These are my observations using the latest firmware version 107a. a) The gateway will register all 4 ports if you're not using password. b) If using password, the gateway will only register the 1st port correctly. For the record, the 2-port 3502 doesn't exhibit this behavior. FYI.

Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty

Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

hello everybody , i got a test setup with lanphone 101 connected to * and a welltech fxo 3802 also to * the extensions are configure so that i can dial from the lanphone to the fxo. although once on the FXO and having the dialtone, no of my dtmf dialing is being processed on the FXO. It keeps giving me the dialtone constatly. Everything is configured as outband DTMF (we

I have a number of Welltech FXS devices (running the SIP code) that I'm trying to register with Asterisk. Has anyone had any success doing this? If I have the Welltech set in peer-to-peer mode, I can call to and from the Welltech, to Asterisk, just fine. However, because it's in peer-to-peer mode, I can only call phone numbers that are in the static phone directory. The phone

I have this in my extension.conf: [incoming_28345474] ; 8862100 is the hotline number of the Welltech 3804 ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Wait(1) exten => 8862100,3,Set(CALLERID(num)=${CALLERID(num)}) include => fax2emailstart [fax2emailstart] exten => 3000,1,SetVar(CALLEDFAX=${EXTEN}) ; me exten => 3000,2,Answer exten =>

Hi there I have some problems in setting up a connection between * and a Wellgate 1501 (1FXS). Either I can dial from * to my Wellgate or I can dial from the Wellgate to *. On the Wellgate you have to setup both "Line number" (which is the number it answers) and "Line account" (which is the account it registers as). Should I expect any problems if these are set the same?

Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff.

https://bugzilla.samba.org/show_bug.cgi?id=13645 Bug ID: 13645 Summary: Improve efficiency when resuming transfer of large files Product: rsync Version: 3.0.9 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: core Assignee:

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or