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@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)