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

[Brian West]

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: 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
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Brian West
> sip:brian at freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Brian West
sip:brian at freeswitch.org



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080425/d0672162/attachment-0002.html