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.
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