You indicate that your 'current attempt' is not working, but you do not say what that involves. Have you defined the SIP trunk 'gateway' into its own XML file within the directory: conf\sip_profiles\external (it is easier to maintain that way) And within the conf\sip_profiles\extenal.xml file have you made sure that the <gateways> will include that new XML file? -------------------------------------------- On Sat, 3/7/15, Richard Mace <richard.mace@gmail.com> wrote: Subject: [Freeswitch-users] SIP Trunk To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Date: Saturday, March 7, 2015, 3:57 AM Hi,I have a trunk that currently works with Asterisk, and I am trying to get it working with FreeSWITCH. The Asterisk config is: [out_trunk]disallow=allhost=sip.voip-unlimited.netusername=usernamefromuser=usernamesecret=passwordtype=peerdtmfmode=rfc2833nat=nocontext=incoming-sipinsecure=inviteallow=alawfromdomain=voip-unlimited.net Any idea how I would configure the same in within FreeSWITCH please, as my current attempt doesn't seem to be working? Thanks very much in advance Richard -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com 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