similar to: Wellgate & Asterisk

I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend

Dear Colleagues, I installed a Wellgate 3804A and overnight lines on all this with frying, putting other lines Wellgate 3804A is well, so I guess it's a problem the first team which is already out of warranty, anyone know how can I fix this? or where to send it in or capital Buenos Aires to fix it? Thanks Mart _________________________________________________________________ Todo

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 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

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

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

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

Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel

Anyone know how to set the wellgate unit so incoming calls pass on directly to asterisk? Right now incoming calls ring twice and I hear a recording saying enter the extension. If I go enter the extension it goes on to asterisk just fine. I just want the incoming call to go directly onto asterisk. Anyone found that out? Jerry

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?

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

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.

Another dumb question... My asterisk system seems to have problems detecting hangups. I am getting a LOT of voicemails with dialtone or silence. I see over at asteriskguru.com there is an explanation of how to configure for polarity reversal in zapata.conf? Does zapata.conf have any function in systems that aren't using zaptel( I suppose not)? I am using an external gateway (wellgate

I have a problem with my Welltech Wellgates. I can't call any extension which starts with or includes * or #. When dialing it responds fine but after some seconds I just get a busy tone and on the Asterisk console it says "SIP/2.0 484 Address Incomplete". Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload

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

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit