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 Linksys WRT54GS router (latest linksys firmware) acting as DHCP and NAT for my home equipment. I have tried forwarding the SIP and RTP ports to the asterisk machine, and I have also tried putting the asterisk machine on the routers DMZ. No luck. I have tried every configuration recommended for stanaphone that I found on the web, including a couple I found on the asterisk wiki and a few I found on the stanaphone forums. I tried, explicitly defining the private and public ips, qualifying, using the same number as assigned in stanaphone for my local extension (a recommendation I found), etc. Then I found a message in a forum about a person with the same problem that claims it got fixed when asterisk was put with a public IP address. So my question is.... is it at all possible to connect asterisk as a SIP client when it sits behind a NAT? If yes, can somebody tell me what I should do please. thank you, -guillermo PS1.- When I connected the x-lite to asterisk both where on the same side of the NAT PS2.- The error I continuosuly get is "SIP/2.0 401 Unauthorized". PS3.- I connected x-lite directly to stanaphone with no problems even behind the nat...and I didn't have to set any port forwarding or anything...so I am thinking that whatever x-lite is doing asterisk should do...how do I emulate what its doing?
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 Linksys WRT54GS router (latest linksys firmware) acting as DHCP and NAT for my home equipment. I have tried forwarding the SIP and RTP ports to the asterisk machine, and I have also tried putting the asterisk machine on the routers DMZ. No luck. I have tried every configuration recommended for stanaphone that I found on the web, including a couple I found on the asterisk wiki and a few I found on the stanaphone forums. I tried, explicitly defining the private and public ips, qualifying, using the same number as assigned in stanaphone for my local extension (a recommendation I found), etc. Then I found a message in a forum about a person with the same problem that claims it got fixed when asterisk was put with a public IP address. So my question is.... is it at all possible to connect asterisk as a SIP client when it sits behind a NAT? If yes, can somebody tell me what I should do please. thank you, -guillermo PS1.- When I connected the x-lite to asterisk both where on the same side of the NAT PS2.- The error I continuosuly get is "SIP/2.0 401 Unauthorized". PS3.- I connected x-lite directly to stanaphone with no problems even behind the nat...and I didn't have to set any port forwarding or anything...so I am thinking that whatever x-lite is doing asterisk should do...how do I emulate what its doing?
I have a problem with SIP on my * box. The * server with a private IP address is behind a NAT modem-router with a public one. I try to connect to a SIP provider which has a * server with public IP but it doesn't works. When I try making a call, the provider answers to the SIP INVITE with a 404 not found error. With a IAX provider, it works find so I am thinking to a NAT problem. Do I have to do port forwarding on my router or to add some special configuration to Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060118/56467332/attachment.htm
On Wed, 2006-01-18 at 15:07 +0100, amaury BOSSE wrote:> I have a problem with SIP on my * box. > > The * server with a private IP address is behind a NAT modem-router > with a public one. > > I try to connect to a SIP provider which has a * server with public IP > but it doesn?t works. > > When I try making a call, the provider answers to the SIP INVITE with > a 404 not found error. > > With a IAX provider, it works find so I am thinking to a NAT problem. > > Do I have to do port forwarding on my router or to add some special > configuration to Asterisk?Yes you have to forward SIP udp port 5060 and RTP udp ports 10000-20000 on your modem to the IP of your Asterisk server. If the port range 10000-20000 is too big for your modem/router than make it something smaller and make sure you change the range also in /etc/asterisk/rtp.conf (rtpstart=10000 & rtpend=20000) Also make sure that you have the following in sip.conf: nat=yes localnet=<your network> (see the examples in sip.conf) Regards, Patrick
Port forwarding might work. My preferred method would be to bridge the connection from the broadband modem to the * box, thus giving it the public IP address. Then add a 2nd NIC to multi-home it and turn it into the router/asterisk/dhcpd/firewall box. Run that connection to a switch and out to your LAN. By doing this you circumvent the SIP/NAT issue. Your Mileage May Vary, Steve Cayona Network Tech Director Super Technologies, Inc.