[Freeswitch-users] Asterisk 1.6 connecting to FS 1.4

[paul.degt at gmail.com]

Hi,
A client of ours is trying to connect his * to our FS, outgoing calls 
work fine, unfortunately when we try to forward an incoming call to his 
* it's not going through.
I see his registration in our internal profile which looks just fine.
We try to forward incoming calls using this in FS dialplan:
<extension name="myext">
      <condition field="destination_number" expression="^555557777777$">   
<action application="bridge" 
data="sofia/internal/4000000@$${domain}|sofia/internal/4000001@$${domain}"/>
      </condition>
    </extension>

Only abnormal things I can see in FS logs are:
2009-09-13 19:17:31.869158 [ERR] sofia_reg.c:1570 No Matching gateway found
2009-09-13 19:17:31.869158 [NOTICE] sofia_reg.c:1590 Hangup 
sofia/internal/4000000[CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]

Why would FS look for a gateway in this case? And what 
MANDATORY_IE_MISSING would mean here? Call gets forwarded to VM as if 
user was unavailable.Hangup is initiated by us in this case.

Client uses this configuration in *:

/etc/asterisk/sip.conf:
>>>>>> /etc/asterisk/sip.conf:

>>>>>> register=>4000000:mysippassword at versafon.com/4000000
>>>>>>
>>>>>> [4000000]
>>>>>> type=friend
>>>>>> username=4000000
>>>>>> secret=mysippassword
>>>>>> host=versafon.com
>>>>>> canreinvite=no
>>>>>> fromuser=4000000
>>>>>> dtmfmode=rfc2833
>>>>>> context=versafon-incoming