Hi there. 
I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as a proxy and registrar ... and freeswitch provides conference calls and voicemail. 
I have calls between two polycoms working and conference calls work. 
But when I try to leave a voice message for a user by dialing 44+ext, the sip proxy never replies to the polycom. 
I did a tcpdump and i can see the INVITE from the polycom to the sip proxy multiple times... but no response back.  The phone eventually disconnects itself. 

Here's what my config looks like:  http://pastebin.com/wWgyVcxc
Just do a search for "route[FSDISPATCH]". 
You will see how I check for the "44" prefix, and then send the call to a route called "FSVBOX".  
Any suggestions would be appreciated. 


  1. route[FSVBOX] {
  2.         if(!($rU=~"^1[0-9][0-9]+$"))
  3.                 return;
  4.         prefix("vb-");
  5.         route(FSRELAY);
  6. }
  7.  
  8. # Send to FreeSWITCH
  9. route[FSRELAY] {
  10.         $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":"
  11.                         + $sel(cfg_get.freeswitch.bindport);
  12.         route(RELAY);
  13.         exit;
  14. }