SIP 404 not found

Hello guyz,
I know this is very common theme and there are many already answered topics, but I can’t solve my problem.
I have installed Asterisk 13.6.0 without DAHDI and libpri.

sip show settings:

Global Settings:
----------------
  UDP Bindaddress:        192.168.50.197:5060
  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:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  No
  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:    No
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 13.6.0
  SDP Session Name:       Asterisk PBX 13.6.0
  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:              asterisk
  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:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  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|h263)
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  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:      No
  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:                public
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Auto (No)
  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:   asterisk

sip.conf:

[home-users](!)
context=homelan
type=user
disallow=all
allow=alaw
allow=ulaw

[ixt01](home-users)
authname=ixt01
secret=Abc12345

[ixt02](home-users)
secret=Abc12345

extensions.conf:

[homelan]
exten => 101,1,Dial(SIP/ixt01)
exten => 102,1,Dial(SIP/ixt02)

log:

<--- SIP read from UDP:192.168.50.125:51823 --->
REGISTER sip:192.168.50.197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.50.125:51823;branch=z9hG4bK-d8754z-b94e84f971806a24-1---d8754z-
Max-Forwards: 70
Contact: <sip:ixt01@192.168.50.125:51823;rinstance=7e4b479938e6dfd1;transport=UDP>
To: <sip:ixt01@192.168.50.197;transport=UDP>
From: <sip:ixt01@192.168.50.197;transport=UDP>;tag=d7487267
Call-ID: ZmFhM2FhODE3YzkxODFkZGQyN2M5NWNmMTkyOGE4NjI.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.50.125:51823 (no NAT)
Sending to 192.168.50.125:51823 (no NAT)

<--- Transmitting (no NAT) to 192.168.50.125:51823 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.50.125:51823;branch=z9hG4bK-d8754z-b94e84f971806a24-1---d8754z-;received=192.168.50.125
From: <sip:ixt01@192.168.50.197;transport=UDP>;tag=d7487267
To: <sip:ixt01@192.168.50.197;transport=UDP>;tag=as6a8631a7
Call-ID: ZmFhM2FhODE3YzkxODFkZGQyN2M5NWNmMTkyOGE4NjI.
CSeq: 1 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

What I’m doing wrong? Or where can I look why it doesn’t find accounts exactly?

PS
This is second try to setup sip.conf :smiley: first time I received not found too, but there is was a record in “sip show subscriptions” and in log was command SUBSCRIBE

Seems that subscribe just for voicemail, when I had set type=friend it was registered immediately.