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: 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: freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@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@freeswitch.org