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 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: [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
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
_______________________________________________
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Brian West
sip:[EMAIL PROTECTED]
_______________________________________________
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
