Well first off you wouldn't use nat.xml for that.. you would clone
default.xml and use it as a base. nat.xml is for OUTBOUND calling from
behind nat only in the default config. its not designed to have
inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasn’t
working. Is there some other place I should have set this?
What’s the difference/application/use of the sample “public” context
versus the “default” one? The sample nat.xml uses the public context.
Thanks,
Jay
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Brian West
Sent: Friday, April 25, 2008 12:01 PM
To: [email protected]
Subject: Re: [Freeswitch-users] Problems with initial setup - basic
nat
You could have just turned auth-calls to false and context to
default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the
authorization errors (for now):
<!-- comment the next line and uncomment one or both of the
other 2 lines for call authentication -->
<param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a
good feature for most people) -->
<param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the
public dialplan with the following:
<extension name="public_call">
<condition field="destination_number" expression="^(.*)$">
<action application="bridge" data="sofia/gateway/gafachi/$1"/>
</condition>
</extension>
Then I wasnt getting 2-way audio so I changed the sip profile for
nat (which Im using internally) and set the ext-sip-ip and the ext-
rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only
using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
<param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the
firewall to our internet sip provider. Obviously the current
configuration isnt secure but its enough to get things going.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Jay Reeder
Sent: Thursday, April 24, 2008 4:40 PM
To: [email protected]
Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30
polycom sip phones. We have an Audiocodes Mediant 2000 to use as a
gateway but for testing I was also trying to setup sip in/out
dialing through the firewall. Ive wanted a reason to start playing
with freeswitch so I thought this would be an excellent opportunity
to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing
something.
I have RC3 on Centos (initially a trixswitch load but then upgraded
to the new RC3) with the standard config files. I did remove the
older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the
outbound sip profile, and the firewalls external ip setup for the
external_rtp and external_sip values (in vars.xml), and the firewall
port forwards all recommended ports(from wiki getting started page)
into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to
Freeswitch on port 5070 (for nat) but Freeswitch wont accept the
call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event
[nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and
authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay
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Brian West
sip:[EMAIL PROTECTED]
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Brian West
sip:[EMAIL PROTECTED]
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