Thanks! :-)
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: [email protected] 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: [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 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 _______________________________________________ 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
