[Freeswitch-users] Problems with initial setup - basic nat

Fri Apr 25 09:40:05 PDT 2008

Aha. Thanks! :-)

We're trying to do outbound calling from behind nat. So the proper
configuration is to still call through the default.xml (port 5060) and it
would call OUT on nat.xml (port 5070)?  In that case, what is outbound.xml
(port 5080) used for?  Would it be for MWI and strictly freeswitch->out
applications?

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, April 25, 2008 12:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat

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!  :-) 

I did have auth-calls set to false in nat.xml but it wasnt working.  Is
there some other place I should have set this?

Whats the difference/application/use of the sample public context versus the
default one?  The sample nat.xml uses the public context.

Thanks,

Jay

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, April 25, 2008 12:01 PM
To: freeswitch-users at lists.freeswitch.org
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: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jay
Reeder
Sent: Thursday, April 24, 2008 4:40 PM
To: freeswitch-users at lists.freeswitch.org
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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