407 Proxy Auth Required on Trunk

Hi all,

I’m seeking some help on a problem I have with Asterisk (Trixbox 2.2). I am using Asterisk to connect to Microsoft Exchange 2007 Unified Messaging, using sipX as a SIP/TCP gateway. I have everything working fine, but for the life of me, cannot work out what is going on in one scenario.

When exchange tries to call an extension to playback a voicemail message, the call fails, with Asterisk responding with a 407 Proxy Auth Required.

Exchange (dc2) is sending the following invite;

<-- SIP read from 192.168.0.50:33974: INVITE sip:400@asterix.lithnet.local;user=phone;transport=udp SIP/2.0 Record-Route: <sip:192.168.0.50:5080;lr;sipX-route=%2Afrom%7ENGFmZjhmNDQ1NQ%60%6 0.authrules%2Aauth%7E%217d8f5d1f3125360fcea92c2d208d19f0> From: <sip:400@dc2.lithnet.local;user=phone>;epid=1A-2D-85-2A-71;tag=4aff8f4455 To: <sip:400@sipx.lithnet.local;user=phone> Cseq: 86 INVITE Call-Id: 9c0761cd-f352-4d3a-afe3-b3ad0ad7a051 Max-Forwards: 18 Via: SIP/2.0/UDP 192.168.0.50:5080;branch=z9hG4bK-5ead2081ce7043fcdcbf81d8c46b72 ea Via: SIP/2.0/UDP 192.168.0.50;branch=z9hG4bK-eb2ba1fca6a851857f79d313de5b7db0 Via: SIP/2.0/TCP 192.168.0.210:46117;branch=z9hG4bK86c66773 Contact: <sip:dc2.lithnet.local:5065;transport=Tcp;maddr=192.168.0.210> Content-Length: 199 User-Agent: RTCC/2.0.6017.0 Content-Type: application/sdp Ms-Conversation-Id: 358e0a19-0565-49b7-a71a-17232fbb36e4 Date: Sun, 05 Aug 2007 15:30:26 GMT

Asterisk is unkindly responding with the following 407;

Using INVITE request as basis request - 9c0761cd-f352-4d3a-afe3-b3ad0ad7a051 Sending to 192.168.0.50 : 5080 (non-NAT) Reliably Transmitting (NAT) to 192.168.0.50:33974: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.50:5080;branch=z9hG4bK-5ead2081ce7043fcdcbf81d8c46b72 ea;received=192.168.0.50 Via: SIP/2.0/UDP 192.168.0.50;branch=z9hG4bK-eb2ba1fca6a851857f79d313de5b7db0 Via: SIP/2.0/TCP 192.168.0.210:46117;branch=z9hG4bK86c66773 From: <sip:400@dc2.lithnet.local;user=phone>;epid=1A-2D-85-2A-71;tag=4aff8f4455 To: <sip:400@sipx.lithnet.local;user=phone>;tag=as02259c9d Call-ID: 9c0761cd-f352-4d3a-afe3-b3ad0ad7a051 CSeq: 86 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26f93896" Content-Length: 0

The sipX Trunk is configured as follows in sip_additional.conf;

[sipXTrunk]
type=friend
host=sipx.lithnet.local
context=from-internal
canreinvite=no
insecure=invite

The extension (400) is configured as follows;

[400] type=friend secret=123456 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes mailbox=400@default host=dynamic dtmfmode=rfc2833 dial=SIP/400 context=from-internal canreinvite=no callerid=device <400>

As far as I can tell from reading the Asterisk manual and various web sites, that should allow it to work. Does anyone have any ideas what I am doing wrong?