similar to: Asterisk and Welltech USB SIP phone K1000A

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

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

Ok this is a simple question... What has been your experience with the WellTech 38xx series (I'm looking specifically at the 3802) VoIP gateway? I'm looking for a good (and hopefully not too expensive) VoIP/T.38 gateway for my office. Asterisk intergration is not a major factor at this time but may be later on. How well does it work? Is Echo a problem? Do the T.38 capablities

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

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 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?

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.

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

We have a samba 3.0.7 server on RHEL-3 (rain) joined as a domain member (security = domain) to a win2k pdc (clouds) for the domain DOM. We have several unix users and two Win-only users. The unix users have matching AD accounts on the win2k, but the Win-only users do not have unix accounts (and we want to keep it that way). So, it seemed that winbind would be the best way to bridge the gap: 1.

Hello list, I need to setup a samba file server with user access from a Windows AD domain and a separate Solaris NIS domain. All of our users have an account on the AD domain but only some of our users have a Unix account. I would like Windows users that have a Unix account to have files written as per their Unix uid and users that do not have an account to have a uid assigned from winbind. I

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

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

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 there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI).

Hi again, Kind of sheepish about asking for help, as I have only spent a day banging my head off this... I got my new Welltech 3701a, 1FXS,1FXO gateway. I flashed it with what is seemingly the appropriate firmware (SIP V1.04). This seems to have gone ok, and it is now registering both ports ok with asterisk. For 1 minute I thought I was home free and and everything was just going to work

Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>