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@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
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