Что сделать чтоб регистрировались пиры (centos/Asterisk 16.8.0)?

Question

[Александр]

После переустановки сервера по данному мануалу pastebin. com /QANKy82r
не регистрируются пиры , статус UNKNOWN, хост Unspecified
в fail2ban апи клиентов добавлены в разрешенные in/out
порты 5060,5061 открыты

Comments

[Viktor]

netstat -lpn | grep asterisk
покажите

[Александр]

Viktor,

udp        0      0 0.0.0.0:5060            0.0.0.0:*                           1649/asterisk       
unix  2      [ ACC ]     STREAM     LISTENING     18247    1649/asterisk        /var/run/asterisk/asterisk.ctl

[Александр]

asterisk -rx "sip show settings"

spoiler

Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  RTP Bindaddress:        Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    Yes
  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          <ip>
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             qwe
  SDP Session Name:       qwe
  SDP Owner Name:         qwe
  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:    Yes
  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:                 (alaw|ulaw|gsm)
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            120 
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  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:No
  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,WS
  Outbound transport:	  UDP
  Context:                all
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        No
  Language:               ru
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk
  RTCP Multiplexing:      No

[Александр]

asterisk -rx "sip show peers"

spoiler

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
101/101                   (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN                                      
102/102                   (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN

[Александр]

sip show peer 101

spoiler

* Name       : 101
  Description  : 
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : internal
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : ru
  Tonezone     : <Not set>
  Accountcode  : internal
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 3
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : invite
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : No
  ContactACL   : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Path support : No
  Path         : N/A
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : (null)
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 101
  SIP Options  : (none)
  Codecs       : (alaw|ulaw|gsm)
  Auto-Framing : No
  Status       : UNKNOWN
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : Yes
  Encryption   : No
  RTCP Mux     : No

[Viktor]

service firewalld stop
service iptables stop
service fail2ban stop
и покажите настройки 101 (что вы там в sip.conf засунули)

[Александр]

101 sip conf

spoiler

[internal](!)
type=friend
host=dynamic
qualify=yes
callgroup=1
pickupgroup=1
callwaiting=no
insecure=invite
;insecure=port,invite
call-limit=3
accountcode=internal
context=internal
canreinvite=no
nat=force_rport,comedia
disallow=all
;allow=g729
allow=alaw
allow=ulaw
allow=gsm


[101](internal)
username=101
secret=123

[Александр]

Viktor, был включен firewalld Спасибо вам огромное!!!