similar to: * <-> FWD behind NAT

<MOD NOTE:Please kill/bounce my other email, it was accidental.> I just pulled down the newest CVS and recompiled. FWD (free world dialup) works now from *, and I AM behind a NAT. I've nearly given up on the xten lite, iaxcomm sounds better. I'll be trying the other win app thats up-and-coming on the list later. It seems to have broken iptel, but that's not as important to

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

My asterisk box is behind a NAT firewall. I have friends that are on Earthlink, Vonage, etc. I'd like to make VOIP calls directly to them rather than going through the PSTN. With Earthlink, I can make this work through FWD peeting numbers, but that's sort of a waste of FWD bandwidth. WIth Vonage, it doesn't work. I suspect this is because of the breakage between FWD and Vonage that

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

--- Peter Zeltins <peter@fintrading.com> wrote: > > > Well, I happen to be one of those very specific cases... ;) and looks > like > will have experiment with it myself. Although I'd hate to re-invent > the > wheel. > > Peter Checking e-mail this morning it looks like we have two independent "fixes" that both do what has been suggested in this

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

It seems if I use port 53, it works (This has both TCP and UDP open) However if I use 500 UDP (TCP is blocked) it doesn't work. On 24 April 2014 19:18, Guus Sliepen <guus at tinc-vpn.org> wrote: > On Thu, Apr 24, 2014 at 07:05:22PM +0100, David Markey wrote: > > > > > Great! How do I enable this feature? > > > > > > It's on by default, just

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK,

Hi all, Love tinc by the way. It's a great VPN. I'm having issues with 2 nodes always talking through an intermediate node. My set up is a VPS in a cloud somewhere that's running tinc and 2 other nodes - one a roaming laptop (always NAT'd) and the other a server behind a dynamic IP home broadband connection (Not NAT'd but firewalled). Neither the laptop nor the home

Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda

Hi all, Here is a graphical diagram of what I am trying to do: <SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone> So I have incoming SIP calls go to the * on the GW, which I then want to forward over IAX to the second * box behind the NAT GW. If I was to place a call on the second * box, it should then forward to the * on the NAT GW

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

Guys. Im trying to figure out how to confgure FWD and NAT. I tried some configs and tested thru the FWD webpage the incoming call.. I got the incoming fine except it kept asking me for my name and seems it didnt get it.. (nat problems, one way sound only)? Also, I cant dialout.. Can somebody using NAT send me some example configs? Thx!

I have noticed that you can't have multiple separate NAT style libvirt networks defined with the same private IP blocks. For example I have this default network: <network> <name>default</name> <uuid>13baf167-02ff-4312-928c-b82ed4df5785</uuid> <forward mode='nat'> <nat> <port start='1024' end='65535'/>

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all