similar to: Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk)

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

I'm just curious why Sipura isn't using free IAX protocol with their devices instead of SIP? With IAX NAT traversal would have been easier, so why are they using SIP. Is there any politics in it? -- #Joseph

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'm experiencing a very unusual problem with a few of my sipuras. They keep over heating after minimal usage. I unplug them and let them sit for 15 minutes and it starts working fine. The other way around is that I put a fan on it or something to keep it cooling. Has anyone else experienced this or are they just defective units? Regards, Mohammed Salim EZZI Telecom, Inc.

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

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 Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

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 can not make a "call pickup" to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? -- #Joseph

Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just

Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura adapters? I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal cards not external units like Linksys. I have one Linksys 3K that with so echo so bad that it is pretty much useless. -- Joseph

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

How to configure Digium S101i adapter to work with "pickupexten *8" ? I have few Sipura adapters and "*8" work OK but my new Digium S101i refuses to cooperate. -- #Joseph

I've a problem with inserting a "pause" and dialing additional numbers when going through Sipura-3000 exten => _12,1,Dial(SIP/4751724@pstn-5,30,D(wwwwww18)) D(wwww) doesn't work as it sends the DTMF tones right after FXS connects to FXO; though, I want insert a "pause" and send additional numbers after connection goes through FXO. Is it possible? -- #Joseph