similar to: Re: Sipura and STUN (was: rejected NOTIFY requests)

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:

When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports: handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx' When I disable NOTIFY messages, * reports the device UNREACHABLE, followed by REACHABLE every couple of minutes. I think I want NOTIFY on, because the Sipura is behind a NAT server, but the constant stream of warnings from * make me think

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

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

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

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

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

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

I need a cheap simple adaptor for analog phones to use with Asterisk. It should be some kind of "configure and forget" type of device, to use at the office, or just throw it in a road warrior's bag and use it while travelling, to call back to the "mothership". I can't decide between iaxy and sipura. Can you guys help? Which one would you use? (and why?) I feel that iaxy

Hi, On my linksys/sipura phones/ATA, there is a setting called "NAT Mapping Enable" and another called "NAT Keep Alive Enable" These settings must be on in my setup so that my phones/ATA remain connected to my * server. My setup is: Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 NAT, Static public IP) - Asterisk server. I was wondering:

Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the order: g729 g711u g711a Any ideas, why the user can hear me, but I cannot hear him (stun) while the other user without stun has no problem. bye Ronald

HI all! I'm looking for some infos to configure stun server support for a SIP peer. I've installed Asterisk 1.4.3, but searching for stun support in chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun... But where i have to put the ip of stun server? No infos around Google and forum! :-) Thank all, regards -- Marco Ciacci Asterisk Admin Windows

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:

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040715/b37cee4b/attachment.htm -------------- next part -------------- ? Hi friends I have some doubt in connecting my firefly3rd party softphone from windows machine to asterisk server in linux . My asterisk is behind the "Port Restricted NAT". I am using STUN server to cross the

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic

I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on