SIP Trunk Authentication Error between Cisco 2951 and FreePBX 13 Using Chan_SIP

Hi,

My organization use Cisco 2951 as voice gateway and Asterisk as internal PBX. This week I met an authentication issue when upgrade my Asterisk&FreePBX to 13.x. outgoing calls were not affected, while all incoming calls come with the error as below.

Debug info from Cisco ISR
Feb 11 14:56:37: //87823/6E36F4E88112/SIP/Error/sipSPIHandleAuthChallenge:
 Error getting credentials
Feb 11 14:56:37: //87823/6E36F4E88112/SIP/Error/act_sentinvite_new_message:
 Error handling AuthenticationChallenge

Here is the configuration from my Cisco 2951 ISR

dial-peer voice 104 voip
 destination-pattern 55733107
 session protocol sipv2
 session target ipv4:10.2.202.60
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw

sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 timers trying 1000
 sip-server ipv4:10.2.202.60

Here is the configuration from my PBX
Asterisk

asterisk-ad*CLI> core show version
Asterisk 13.7.0 built by root @ asterisk-ad on a x86_64 running Linux on 2016-01-29 09:54:08 UTC

asterisk-ad*CLI> sip show settings


Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5061
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Path support :          No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-13.0.54(13.7.0)
  SDP Session Name:       Asterisk PBX 13.7.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              Unknown
  From: Domain:          
  Record SIP history:     Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 (ulaw|alaw|gsm|g726)
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            30
  RTP Hold Timeout:       300
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      Yes
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:        UDP
  Context:                from-sip-external
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        No
  Language:               en
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   *97

SIP Trunks
[2951]
type=friend
context=from-trunk
host=192.168.245.1
dtmfmode=rfc2833
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=yes

[AsteriskNow-qd-2911]
context=from-internal
host=192.168.245.1
type=friend
qualify=yes
dtmfmode=rfc2833
allow=ulaw
allow=alaw
nat=no
insecure=port,invite


Please help review and let me any mistakes I made

Thanks,
Chandler