similar to: Connecting SPA-300 to Asterisk

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

Dear Expert, I am stuck when trying to register SPA-3102 on AsteriskNow .. could any body please advise .. where can I find the article for doing this? .. I googled but got nothing.. Regards bie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071203/6bbbecd8/attachment.htm

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>

I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Anyone know what I need to do to get the FXO port on the SPA 3000 to forward calls to Asterisk? My Asterisk is running on port 5061 and I set the dial plan on the device to forward to s@asteriskip:5061 but Asterisk is not picking it up. I can see on tcpdump traces that the Invite packets do go to through to the asterisk machine on port 5061, but it's not picking them up. sip debug does not

Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ? From a quick test (got mine yesterday), seems like it is not recognizing Caller ID from PSTN/FXO port.. Using the same configuration as a Sipura 3000 (to be sent to mother-in-law POP :-), no Caller ID at all, (I've even extended the PSTN delay to give it more time, but no dice). www.voxilla.com forum has a couple