asterisk users - Jan 2011 - cannot answer incoming calls

Have recently installed some Snom phones into an office. Phones are
natted and connect to a 1.4 server on a public IP

We can make outgoing calls, but are unable to answer incoming calls.
The phone rings, but the call cannot be picked up. Other phones on
other sites connected to the server are working perfectly.

Looking at the SIP trace it appears the phone transmits:

Sent to udp:193.33.xx.xx:5060 at 6/1/2011 11:49:20:868 (849 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 193.33.xx.xx:5060;branch=z9hG4bK6e82052c;rport=5060
From: "xx"
<sip:07765000000 at sip3.office-voip.com>;tag=as1b6fc27c
To: <sip:xxxxx_203 at 79.123.xx.xx:25380>;tag=37gg1zu3wp
Call-ID: 1b212085091e98387237125f0ab810fa at sip3.office-voip.com
CSeq: 102 INVITE
Contact: <sip:xxxxx_203 at 192.168.4.19:2048>;reg-id=1
User-Agent: snom300/7.3.30
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 641540583 641540584 IN IP4 192.168.4.19
s=call
c=IN IP4 192.168.4.19
t=0 0
m=audio 52386 RTP/AVP 18 101
a=rtpmap:18 g729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

but it is never received by the server.

Interestingly RINGING and REGISTER messages are working OK. The NAT
router is out of our control. Are we looking at a SIP ALG getting in
the way?

Thanks,

John

Hi John,
> Interestingly RINGING and REGISTER messages are working OK. The NAT
> router is out of our control. Are we looking at a SIP ALG getting in
> the way?
It probably is the NAT router. Have you tried "canreinvite=no" in
sip.conf for these phones?

Best regards,
Jeroen Eeuwes