similar to: OT: Opensource "Sipura Profile Compiler" for SPA2K, 3K

James H. Thompson wrote: > The Sipuras pull their config from a HTTP or TFTP server. > Now that they support XML config files, all you need to do is put the > file on your web server and point the sipura to it. > The Sipura will pull down the config from your web server, nothing > special required on the web server side. > And only a single line entry on the Sipur

All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to

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

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?

I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER <sip:714XXXXXXX@1.0.24.5>;tag=7f8072c0c46250f7o1 X's are there to

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

<< Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits.>> I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 10000 - 20000 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph

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

The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER <sip:A714XXXXXXX@1.0.24.5>;tag=1a6833c3913bcb6o1

Does anyone have a Sipura 3k running, and using the FXO? I've got things working right, but if I try to toss a *67 in the dialplan, it seems the sipura is throwing a 403 forbidden back. For example: exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/${EXTEN:1} works fine exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/*67${EXTEN:1} does not (even if I toss a couple Ws in) I can't

I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

Hi all, Am looking for everyones advise/recommendations. I have am setting up a network of both office and home based workers. The office workers will be on the same network as the Asterisk box so no NAT hassles there. However, the home workers are on their own DSL connections so I imagine that most will be behind NAT. I had hoped to use Sipura SPA-2000's but I presume I would need either

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys