similar to: Asterisk -> Sipura SPA3000 peer behind NAT

Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor

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

I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message:

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

Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? If you have an idea, please tell me! Thanks, Micka?l

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Hi, I'm using a SPA 3000 as FXO and FXS termination connected to my * box. I'm using the caller ID prefix trick explained here: http://www.voip-info.org/wiki-Sipura+3000. All seems to work really fine, there is only a problem when I hangup my IP phone after a conversation and the other party don't: in this case the PSTN line remains connected until the other party hangs up. Any

Hi All, I'm on of those brave souls who bought into the preproduction beta of the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and am exploring it's workings. I really want it mostly as a straightforward FXO adapter, to replace an X101p. Let me be clear, I'd love to support Digium in every way possibe, and will likely buy a TDM40 card shortly. But, the X101p has

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

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet. It works fine to my local A@H box. I've tried... many things. I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working. Any Sipura experts out there? Eric.

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

I'm new to asterisk and already a fan. Please forgive me if my questions are covered by some FAQ and thanks in advance for any pointers anyone can give me. The basic problem that I'm having is that sometimes outgoing calls result in a busy signal when the outgoing line is free. I'm thinking that the channel is timing out or something but haven't figured out how to debug or gather

Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of

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

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

In trying to get my services up and running, I've encountered the usual spate of first-time issues. I was wondering if there was a good FAQ or Howto on troubleshooting NAT issues. The equipment that I'm using is typically a Sipura ATA or hardphone (SPA-942) sitting behind either a Linux box as firewall (using IPtables and Arno's firewall) or else a Cisco IOS device running 12.4T