similar to: zoom FXS/FXO gateways

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

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

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

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, 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

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

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Has anyone configured the Zoom V3 to connect with Asterisk? I bought one at Fry in San Diego for $99 bucks. I can't get it to register with Asterisk.. Thanks, David http://www.zoom.com/products/voip_products.html

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they "stole" from me. I only bought from them because they showed a 4-day availability stock level! Now

Hello All! I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000. The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk. SPA HTTP Configuration:

I'm looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/d66e6335/attachment.htm

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