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