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opensips with asterisk integration question


I am trying to conigured opensips as registar and asterisk for media
services using this doc:
https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration

so far everything went well and i am able to use VoiceMail, MeetMe
etc. but when i am trying to dial any SIP phone getting following
error

Asterisk logs:

[Jul 17 15:23:05] NOTICE[9933][C-00000001]: chan_sip.c:31847
build_peer: The 'username' field for sip peers has been deprecated in
favor of the term 'defaultuser'
[Jul 17 15:23:05] WARNING[9933][C-00000001]: sip/config_parser.c:817
sip_parse_nat_option: nat=yes is deprecated, use
nat=force_rport,comedia instead
[Jul 17 15:23:05] WARNING[9972][C-00000001]: sip/config_parser.c:817
sip_parse_nat_option: nat=yes is deprecated, use
nat=force_rport,comedia instead
[Jul 17 15:23:05] NOTICE[9933][C-00000001]: chan_sip.c:24288
handle_response_invite: Failed to authenticate on INVITE to '"1001"
<sip:1001 at opensips.org>;tag=as1986a977'
voip*CLI>

In mysql sipusers look like following:

*************************** 3. row ***************************
       name: 1001
   username: 1001
       type: friend
     secret: NULL
       host: opensips.org
   callerid: NULL
    context: default
    mailbox: 1001
        nat: yes
    qualify: no
   fromuser: 1001
   authuser: NULL
 fromdomain: opensips.org
   insecure: NULL
canreinvite: no
   disallow: NULL
      allow: NULL
restrictcid: NULL
  defaultip: opensips.org
     ipaddr: opensips.org
       port: 5060
 regseconds: NULL

basis dialplan

; announcement demo
exten => 2000,1,Ringing
exten => 2000,2,Playback(welcome)
exten => 2000,3,Hangup

; voicemail
exten => 777,1,Ringing
exten => 777,n,Wait(1)
exten => 777,n,Answer
exten => 777,n,Wait(1)
exten => 777,n,VoiceMailMain(@default)
exten => 777,n,Hangup

exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)