opensips with asterisk integration question
Paul Belanger
pabelanger at redhat.com
Thu Jul 18 13:16:49 UTC 2019
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)
>
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