[Freeswitch-users] SIP Trunk

[T Fred Farmington]

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 at gmail.com> wrote:

 Subject: [Freeswitch-users] SIP Trunk
 To: "FreeSWITCH Users Help" <freeswitch-users at 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 at 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 at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org