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