Hi all, I have solved the above problem of routing call from SIP client (Yate) to Freeswitch: All the configuration was same as my previous request, but instead of creating a local user and gateway, i only created one gateway in directory/default folder with all the settings like (used <include> <user id="4444"> <gateways> <gateway name="yate"> <param name="username" value="4444"/> <param name="realm" value="192.168.1.23"/> <param name="password" value="''your-password''"/> <param name="proxy" value="192.168.1.23"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </gateways> <params> <param name="password" value="1234"/> </params> </user> </include> Enjoy; Shehzad.