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opensips with asterisk integration question


On Wed, Jul 17, 2019 at 10:28:14PM -0400, Satish Patel wrote:
> I am trying to conigured opensips as registar and asterisk for media
> services using this doc:
> https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration
> 
Greeting! You are currently on the OpenStack mailing list, you likely
want to use:

  http://lists.digium.com/mailman/listinfo/asterisk-users

> 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
> 
These aren't errors, but warnings and notice. It shouldn't affect
anything, but asterisk is suggesting you make changes.

> 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>
> 
Looking at your config below, I don't see you setting a secret. It is
possible your opensips server is expecting one. If not, I would set one
to help avoid unathenticated calls.

> 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)
>