[Freeswitch-users] Asterisk registration with FS

[Noah Silverman]

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 at 127.0.0.1
User            3235551212 at 111.111.111.111
Contact         "user" <sip: 3235551212 at 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 at 10.0.1.110
User            3235551212 at 111.111.111.111
Contact         "user" <sip: 3235551212 at 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