Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your Asterisk@Home Server via SIP through a NAT ) To be precise i did the following (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 10000 to 20000 to 192.168.1.2 (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0 (3) I also set nat=yes qualify=yes (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your Asterisk@Home Server via SIP through a NAT ).Please advise me how to make it work !!! (5) I am using xten lite soft phone on my pc . (6) I use cisco 1700 series router ,and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,10000 to 20000 to 192.168.1.2 on a cisco router . Please reply and advice !!! Thanks --------------------------------- Yahoo! Photos ? Showcase holiday pictures in hardcover Photo Books. You design it and we?ll bind it! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060112/96ad4471/attachment.htm
Please note that recent IOS has SIP NAT traversal turned on by default. I believe that it only supports internal UA / external server. Since you also want the opposite, you should probably turn it off: no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060 Some IOS versions will even crash on SIP behind NAT. See http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. You can edit rtp.conf to use e.g 10000-10007 (would allow 4 calls) and then only 8 ip nat statements would be needed for RTP. You don't say what's failing. "make calls outside our LAN" sounds like you are trying to call using a VoIP provider that Asterisk registers with. But "your remote SIP phones" is something different; which of the above are failing? Are the registrations successful? Is it just the RTP that's not working (in which case the called phone will still ring)? If not, what error or timeout is reported? If * verbose and/or debug logs don't show precisely what is going wrong, use Ethereal (on both sides of the router if necessary) to see what is happening. --Stewart> Hi all , > I have tried configuring Asterisk at home to make calls outside our Lan > WITHOUT any success (Setting up your router/firewall so your remote SIP > phones can communicate with your Asterisk@Home Server via SIP through a > NAT ) > > To be precise i did the following > > (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 > Forward UDP Port 10000 to 20000 to 192.168.1.2 > > (2) I set externip = x.x.x.x (to our public WAN) > localnet =192.168.1.0 /255.255.255.0 > > (3) I also set nat=yes > qualify=yes > > (4)Please,I know alot of you out there have implemented AAH to work > outside your network ( Setting up your router/firewall so your remote SIP > phones can communicate with your Asterisk@Home Server via SIP through a > NAT ).Please advise me how to make it work !!! > > (5) I am using xten lite soft phone on my pc . > > (6) I use cisco 1700 series router ,and i have natting configured on > this router .Maybe I am using a wrong command .Please,tell me the > commands to forward the ports Port 5060-5082,10000 to 20000 to > 192.168.1.2 on a cisco router . > > Please reply and advice !!! > Thanks
Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. just add the range ports tih a ":" e.g 192.168.1.2 10000 : 10007 > (4)Please,I know alot of you out there have implemented AAH to work > outside your network ( Setting up your router/firewall so your remote SIP > phones can communicate with your Asterisk@Home Server via SIP through a > NAT ).Please advise me how to make it work !!! If what you are trying to do is a SIP --> NAT --> Internet --> Nat --> Asterisk call them I'm afraid you would need to use a SIP/RTP router. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Thu Jan 12 09:29:42 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Please note that recent IOS has SIP NAT traversal turned on by default. I believe that it only supports internal UA / external server. Since you also want the opposite, you should probably turn it off: no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060 Some IOS versions will even crash on SIP behind NAT. See http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. You can edit rtp.conf to use e.g 10000-10007 (would allow 4 calls) and then only 8 ip nat statements would be needed for RTP. You don't say what's failing. "make calls outside our LAN" sounds like you are trying to call using a VoIP provider that Asterisk registers with. But "your remote SIP phones" is something different; which of the above are failing? Are the registrations successful? Is it just the RTP that's not working (in which case the called phone will still ring)? If not, what error or timeout is reported? If * verbose and/or debug logs don't show precisely what is going wrong, use Ethereal (on both sides of the router if necessary) to see what is happening. --Stewart> Hi all , > I have tried configuring Asterisk at home to make calls outside our Lan > WITHOUT any success (Setting up your router/firewall so your remote SIP > phones can communicate with your Asterisk@Home Server via SIP through a > NAT ) > > To be precise i did the following > > (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 > Forward UDP Port 10000 to 20000 to 192.168.1.2 > > (2) I set externip = x.x.x.x (to our public WAN) > localnet =192.168.1.0 /255.255.255.0 > > (3) I also set nat=yes > qualify=yes > > (4)Please,I know alot of you out there have implemented AAH to work > outside your network ( Setting up your router/firewall so your remote SIP > phones can communicate with your Asterisk@Home Server via SIP through a > NAT ).Please advise me how to make it work !!! > > (5) I am using xten lite soft phone on my pc . > > (6) I use cisco 1700 series router ,and i have natting configured on > this router .Maybe I am using a wrong command .Please,tell me the > commands to forward the ports Port 5060-5082,10000 to 20000 to > 192.168.1.2 on a cisco router . > > Please reply and advice !!! > Thanks_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060112/0f497eea/attachment.htm
Thanks .Find My replies in between your lines
"Please note that recent IOS has SIP NAT traversal turned on by default.
I believe that it only supports internal UA / external server.
Since you also want the opposite, you should probably turn it off:
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
Some IOS versions will even crash on SIP behind NAT. See
http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html?
Hope your comment above will not affect the Natting configuration on my router .
What will be the effect of me turning it off ?
?You don't say what's failing. "Make calls outside our LAN"
sounds like
you are trying to call using a VoIP provider that Asterisk registers
with. But "your remote SIP phones" is something different; which of
the above are failing? Are the registrations successful? Is it just
the RTP that's not working (in which case the called phone will still
ring)? If not, what error or timeout is reported??
We are not using a VOIP service provider, we use asterisks server behind a nat
device (Cisco router), and the asterisk server is connected to an E1 link. We
can make and receive calls on pc with xten that is on the same private LAN with
the asterisk server
?If * verbose and/or debug logs don't show precisely what is going
wrong,
use Ethereal (on both sides of the router if necessary) to see what
is happening.?
How can I use verbose and/or debug logs and Ethereal
---------------------------------
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