Hi, I have our FS server configured and running. I can register a polycom phone to it perfectly. Can make calls out and receive calls in. Works well, quality is solid, etc. Now, I want to register an asterisk box with FS. (Our eventual plan is to resell VOIP as a "trunk" to end users, so this is an important test.) FS won't route the calls from asterisk. I don't see any specific errors in the FS console, but the calls just die. Below is the result of "sofia status profile default" showing that both the asterisk box and polycom phone did successfully register. (username, password, host changed for privacy) Call-ID 456fa3406d67ee337c6c81264932f76a@127.0.0.1 User 3235551212@111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:1024;fs_nat=yes> Agent Asterisk PBX Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 12:17:40) Call-ID 8c5f3b02-c1f07a54-7d8ac0c7@10.0.1.110 User 3235551212@111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:5060;fs_nat=yes> Agent PolycomSoundPointIP-SPIP_500-UA/2.1.3.0028 Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 13:57:15) Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy) [Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw Can anyone help me figure out what is wrong? Thanks, -N