Anyone use a virtual number with Asterisk ?

I have a virtual number and when I get an incoming call, I am getting two INVITE messages. Voipo sends a cancel for the first INVITE and is sending a second.

<--- SIP read from 174.37.45.134:5060 --->
INVITE sip:4085555555@79.23.170.73 SIP/2.0^M
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
f: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
t: <sip:+12125555555@67.231.8.93:5060>^M
i: FOOBAR@209.244.63.25^M
CSeq: 1 INVITE^M
m: <sip:+12015555555@4.68.250.148:5060;transport=udp; nat=yes>^M
Max-Forwards: 64^M
c: application/sdp^M
Content-Length: 285^M
P-Asserted-Identity: "AnonUser" <sip:2015555555@pstn>^

<--- Transmitting (NAT) to 174.37.45.134:5060 --->
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0;receiv ed=174.37.45.134^M
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
From: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
To: <sip:+12125555555@67.231.8.93:5060>^M
Call-ID: FOOBAR@209.244.63.25^M
CSeq: 1 INVITE^M
User-Agent: Asterisk PBX^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
Supported: replaces, timer^M
Contact: <sip:4085555555@79.23.160.53>^M
Content-Length: 0^

Asterisk processes the call

<--- SIP read from 174.37.45.134:5060 --->
CANCEL sip:4085555555@79.29.140.37 SIP/2.0^M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
From: "AnonUser" <sip:+12015555555@4.68.250.148>;tag=VPSF5060716294 60^M
Call-ID: FOOBAR@209.244.63.25^M
To: <sip:+12125555555@67.231.8.93:5060>^M
CSeq: 1 CANCEL^M
Max-Forwards: 70^M
User-Agent: Kamailio (1.4.3-notls (x86_64/linux))^M
Content-Length: 0

<--- Reliably Transmitting (NAT) to 174.37.45.134:5060 --->
SIP/2.0 487 Request Terminated^M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0;receiv ed=174.37.45.134^M
From: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
To: <sip:+12125555555@67.231.8.93:5060>;tag=as47f04e2b ^M
Call-ID: FOOBAR@209.244.63.25^M
CSeq: 1 INVITE^M
User-Agent: Asterisk PBX^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
Supported: replaces, timer^M
Content-Length: 0

<--- SIP read from 174.37.45.134:5060 --->
ACK sip:4085555555@79.29.150.39 SIP/2.0^M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
From: "AnonUser" <sip:+12015555555@4.68.250.148>;tag=VPSF5060716294 60^M
Call-ID: FOOBAR@209.244.63.25^M
To: <sip:+12125555555@67.231.8.93:5060>;tag=as47f04e2b ^M
CSeq: 1 ACK^M
Max-Forwards: 70^M
User-Agent: Kamailio (1.4.3-notls (x86_64/linux))^M
Content-Length: 0

<--- SIP read from 174.37.45.134:5060 --->
INVITE sip:4085555555@79.28.129.39 SIP/2.0^M
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
Allow: INVITE,ACK,CANCEL,BYE^M
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.9c6816c4.0^M
f: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
t: <sip:+12125555555@67.231.8.93:5060>^M
i: FOOBAR@209.244.63.25^M
CSeq: 1 INVITE^M
m: <sip:+12015555555@4.68.250.148:5060;transport=udp; nat=yes>^M
Max-Forwards: 64^M
c: application/sdp^M
Content-Length: 285^M
P-Asserted-Identity: "AnonUser" <sip:2015555555@pstn>