SIP Phone outgoing calls

I have a SNOM 300 and I can receive calls both by internal extensions and external callers via Google Voice/MOTIF, however I cannot make ANY calls. However using a soft phone both on my PC and on my IPhone I can make calls no problem. Im thinking its something with the way the phone handles SIP. Any help would be greatly appreciated. Here is my SIP log from the phone.

Sent to udp:192.168.2.135:5060 at 8/9/2013 20:02:19:272 (1142 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-cutclt8p2znj;rport
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[email protected]:2049;reg-id=1
X-Serialnumber: 00041328E38F
P-Key-Flags: keys="3"
User-Agent: snom300/8.5.8-OCS
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, BENOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 407

v=0
o=root 340582393 340582393 IN IP4 192.168.2.132
s=call
c=IN IP4 192.168.2.132
t=0 0
m=audio 60066 TCP/RTP/AVP 0 8 3 106 18 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:106 g726-32/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=connection:new
a=setup:active
a=sendrecv


Received from udp:192.168.2.135:5060 at 8/9/2013 20:02:19:351 (521 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-cutclt8p2znj;received=192.168.2.132;rport=2049
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0eb5f58f"
Content-Length: 0


Sent to udp:192.168.2.135:5060 at 8/9/2013 20:02:19:361 (360 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-cutclt8p2znj;rport
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 1 ACK
Max-Forwards: 70
Contact: sip:[email protected]:2049;reg-id=1
Content-Length: 0


Sent to udp:192.168.2.135:5060 at 8/9/2013 20:02:19:378 (1311 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-vvb0yk6zm4bi;rport
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 2 INVITE
Max-Forwards: 70
Contact: sip:[email protected]:2049;reg-id=1
X-Serialnumber: 00041328E38F
P-Key-Flags: keys=“3"
User-Agent: snom300/8.5.8-OCS
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, BENOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username=“101”,realm=“asterisk”,nonce=“0eb5f58f”,uri="sip:[email protected];user=phone”,response=“bb85c05f928e14ffe04bf7d5dfef34b5”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 407

v=0
o=root 340582393 340582393 IN IP4 192.168.2.132
s=call
c=IN IP4 192.168.2.132
t=0 0
m=audio 60066 TCP/RTP/AVP 0 8 3 106 18 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:106 g726-32/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=connection:new
a=setup:active
a=sendrecv


Received from udp:192.168.2.135:5060 at 8/9/2013 20:02:19:407 (521 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-cutclt8p2znj;received=192.168.2.132;rport=2049
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0eb5f58f"
Content-Length: 0


Sent to udp:192.168.2.135:5060 at 8/9/2013 20:02:19:411 (360 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-cutclt8p2znj;rport
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 1 ACK
Max-Forwards: 70
Contact: sip:[email protected]:2049;reg-id=1
Content-Length: 0


Received from udp:192.168.2.135:5060 at 8/9/2013 20:02:19:417 (452 bytes):

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-vvb0yk6zm4bi;received=192.168.2.132;rport=2049
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Sent to udp:192.168.2.135:5060 at 8/9/2013 20:02:19:427 (360 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:2049;branch=z9hG4bK-vvb0yk6zm4bi;rport
From: “EBO” sip:[email protected];tag=wy44lpjc3r
To: sip:[email protected];user=phone;tag=as7f8acde8
Call-ID: 3c26716d19a6-7dgr8s04j9m1
CSeq: 2 ACK
Max-Forwards: 70
Contact: sip:[email protected]:2049;reg-id=1
Content-Length: 0

Turn off secure RTP on the phone.

Thanks…that worked perfect. This forum rocks. Solved all my issues in less than 24 hours!