Inbound calls "Proxy Authentication Required"

[rwfaught]

A few weeks ago we set up a new DID for the purpose of playing back a message to callers. I set it up and we had no problems for a couple of weeks. Then outta nowhere calls to the DID started ringing 2-3 times then a fast busy signal. Nothing had changed on our configuration to my knowledge but the carrier said they are getting back a 407 from us - we are not authenticating the call. This is the same carrier we are placing outbound calls to with no problems. So I look at sip debug and see my call coming in and sure enough I see "Proxy Authentication is required". I'm afraid I might have inadvertently done something and forgot about it. Can you guys tell me where I should look next?


<------------->
[Apr 5 11:48:31] --- (11 headers 12 lines) ---
[Apr 5 11:48:31] Found RTP audio format 18
[Apr 5 11:48:31] Found RTP audio format 101
[Apr 5 11:48:31] Found audio description format G729 for ID 18
[Apr 5 11:48:31] Found audio description format telephone-event for ID 101
[Apr 5 11:48:31] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[Apr 5 11:48:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 5 11:48:31] Peer audio RTP is at port 152.184.0.110:12112
[Apr 5 11:48:31]
<--- SIP read from x.x.x.x:5060 --->
INVITE sip:417XXXXXXX@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK1sansay133293794rdb4671
Record-Route: <sip:sansay133293794rdb4671@xx.xx.xx.xx:5060;lr;transport=udp>
To: <sip:417XXXXXXX@xx.xx.xx.xx>
From: <sip:1417XXXXXXX@xx.xx.xx.xx>;tag=sansay133293794rdb4671
Call-ID: 76032939-0-300117920@xx.xx.xx.xx
CSeq: 1 INVITE
Contact: <sip:1417XXXXXXX@xx.xx.xx.xx:5060>
Supported: timer,100rel
Session-Expires: 1800;refresher=uac
Min-SE: 90
Remote-Party-ID: <sip:417XXXXXXX@xx.xx.xx.xx:5060>;privacy=off
Max-Forwards: 66
Content-Type: application/sdp
Content-Length: 273

v=0
o=Sansay-VSXi 188 1 IN IP4 xx.xx.xx.xx
s=Session Controller
c=IN IP4 208.93.227.5
t=0 0
m=audio 13440 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20[Apr 5 11:48:31] --- (15 headers 13 lines) ---
[Apr 5 11:48:31] Sending to x.x.x.x : 5060 (NAT)
[Apr 5 11:48:31] Using INVITE request as basis request - 76032939-0-300117920@x.x.x.x
[Apr 5 11:48:31] Found peer 'sip'
[Apr 5 11:48:31]
<--- Reliably Transmitting (NAT) to x.x.x.x:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK1sansay133293794rdb4671;received=xx.xx.xx.xx
From: <sip:1417XXXXXXX@xx.xx.xx.xx>;tag=sansay133293794rdb4671
To: <sip:417XXXXXXX@xx.xx.xx.xx>;tag=as69bb325f
Call-ID: 76032939-0-300117920@xx.xx.xx.xx
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="258336f2"
Content-Length: 0

[rwfaught]

After looking at some posts about this issue on these forums I asked my carrier about the possibility that we are not authenticating because of an IP address change. They informed me that they installed a new session border controller about 2-3 weeks ago which would be the same time this started happening. I have sent them the SIP debug info and they are looking it over but they tell me that I need to change my configuration to all their 29 bit network access. I'm hoping this will be resolved without me having to make changes to my dialer.