Hi All
Yes I know double Nat is a problem
But I have a Cisco 7960 which is remote from the * PBX ad connected via
the Internet. Each side has NAT
(1) Sometimes it will work often it won't. And when it decides to work
is random
Always
(2) The Register side works fine. SIP SHOW PEERS has the phone listed
with the correct IP address and an average Qualify time (121 ms)
Always
(3) You can make calls outbound with the Cisco phone through the * PBX
Problem
(4) You can not receive any calls (when not working correctly)
(a) The Phone rings but not voice goes through
(b) Sometimes get a 481 "Call Leg Does Not Exist"
(c) Sometimes get a -- is circuit-busy
(5) On a reload of the * box you will 95 % sure loose the connection if
it was working ?
(6) SIP 5060 - 5063 and RTP 10000 - 25000 is open and port forwarded
on both sides
(7) All calls are VoIP and terminate or originate via a VoIP Provider
Anybody got any ideas, I have tried everything
Thanks All
Barry
Barry Fawthrop wrote:> Hi All > > Yes I know double Nat is a problem > > But I have a Cisco 7960 which is remote from the * PBX ad connected > via the Internet. Each side has NAT > > (1) Sometimes it will work often it won't. And when it decides to work > is random > > Always > (2) The Register side works fine. SIP SHOW PEERS has the phone listed > with the correct IP address and an average Qualify time (121 ms) > > Always > (3) You can make calls outbound with the Cisco phone through the * PBX > > Problem > (4) You can not receive any calls (when not working correctly) > (a) The Phone rings but not voice goes through > (b) Sometimes get a 481 "Call Leg Does Not Exist" > (c) Sometimes get a -- is circuit-busy > > (5) On a reload of the * box you will 95 % sure loose the connection > if it was working ? > > (6) SIP 5060 - 5063 and RTP 10000 - 25000 is open and port > forwarded on both sides > > (7) All calls are VoIP and terminate or originate via a VoIP Provider > > Anybody got any ideas, I have tried everything > > Thanks All > Barry >You could try giving up and not wasting anymore time. At least that was my experience after spending MANY hours working on a solution. Well I came up with a solution, and it was to remove the double NAT, at least to layer 3 of the stack. OpenVPN saved the day. Thanks, Steve Totaro
On the 7960 with a SIP image, Press the <Settings> button and go to
option 4 "SIP Configuration". Scroll down to line "24 NAT
Enabled" and
set it to yes. Then set "25 NAT Address" to the external IP address.
This will need to be manually changed every time the phone's router
pulls a new DHCP lease. In your sip.conf, make sure that you have
nat=yes and qualify=yes. I have had double-NATed 7960s work with this
setup, but you are at the mercy of the routers involved in performing
the NAT.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Barry
Fawthrop
Sent: Sunday, September 24, 2006 5:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco 7960 Double Natted
Hi All
Yes I know double Nat is a problem
But I have a Cisco 7960 which is remote from the * PBX ad connected via
the Internet. Each side has NAT
(1) Sometimes it will work often it won't. And when it decides to work
is random
Always
(2) The Register side works fine. SIP SHOW PEERS has the phone listed
with the correct IP address and an average Qualify time (121 ms)
Always
(3) You can make calls outbound with the Cisco phone through the * PBX
Problem
(4) You can not receive any calls (when not working correctly)
(a) The Phone rings but not voice goes through
(b) Sometimes get a 481 "Call Leg Does Not Exist"
(c) Sometimes get a -- is circuit-busy
(5) On a reload of the * box you will 95 % sure loose the connection if
it was working ?
(6) SIP 5060 - 5063 and RTP 10000 - 25000 is open and port forwarded
on both sides
(7) All calls are VoIP and terminate or originate via a VoIP Provider
Anybody got any ideas, I have tried everything
Thanks All
Barry
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