I am using embedded freeswitch as a softphone client and I am trying to subscribe to call-info on the server, (see config below) but the server is responding with a 481 Call/Transaction Does not exist. I compared the freeswitch SIP messages with SIP messages sent by a polycom phone for this feature and I noticed that freeswitch doesn't send the username in the request line. I think that this is causing the 481 response. Polycom Version: SUBSCRIBE sip:user@server:5060;transport=udp SIP/2.0 Freeswitch: SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 Below is my gateway configuration <gateway name='1'> <param name='realm' value='SIP_SERVER' /> <param name='username' value='SIP_USERNAME' /> <param name='password' value='SIP_PASSWORD' /> <param name='register' value='true' /> <param name='extension' value='SIP_USERNAME' /> <param name='extension-in-contact' value='true' /> <subscriptions name='x'> <subscription event='call-info'> <param name='retry-seconds' value='40'/> <param name='expire-seconds' value='3600'/> <param name='content-type' value=''/> </subscription> </subscriptions> </gateway> Is there any way to tell freeswitch to include the username in the request line? Thank You, Duvid Rottenberg