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

[Jay Reeder]

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 wasn't getting 2-way audio so I changed the sip profile for nat
(which I'm 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 I'm 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 isn't
secure but it's 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

 

We're 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.
I've wanted a reason to start playing with freeswitch so I thought this
would be an excellent opportunity to use freeswitch for the Nat traversal.

 

I've been through the wiki and reviewed list archives but I'm 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 firewall's 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 I'm stuck.  I have sipx attempting to send calls to Freeswitch
on port 5070 (for nat) but Freeswitch won't 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.  

 

I'm sure it's something obvious but what am I missing?

 

Thanks,

 

Jay

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