asterisk.ru

Код: Выделить всё

[general]
register => +375XXXXXXXX@ims.beltel.by:QWERTY:"+375XXXXXXXX@ims.beltel.by"@10.24.0.41/+375XXXXXXXX


[ims222]
defaultuser=+375XXXXXXXX@ims.beltel.by
type=peer
secret=QWERTY
qualify=yes
nat=force_rport,comedia
insecure=invite,port
host=10.24.0.41
fromuser=+375XXXXXXXX
fromdomain=ims.beltel.by
dtmfmode=inband
disallow=all
context=ims_incoming
allow=g722,alaw,ulaw

Код: Выделить всё

sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
ims222/+375XXXXXXXX@ims. 10.24.0.41                                  Yes        Yes            5060     OK (15 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
10.24.0.41:5060                         N      +375XXXXXXXX       105 Registered           Thu, 22 Mar 2018 20:53:12
1 SIP registrations.

*CLI> core show version
Asterisk 13.1.0~dfsg-1.1ubuntu4.1 built by buildd @ lgw01-12 on a i686 running Linux on 2017-04-04 11:28:44 UTC

*CLI> sip show settings
Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  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:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  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:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
  SDP Session Name:       Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
  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:         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 403:0
  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:                default
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Auto (No)
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk

Код: Выделить всё

[general]                                     
static=yes                                    

[globals]                                     

[ims_incoming]                                
exten => +375XXXXXXXX,1,Answer()             
exten => s,n,Wait(1)                          
exten => s,n,NoOp(1)                          
                                              
exten => s,n,Playback(hello-world)            
exten => s,n,Hangup()

Код: Выделить всё

*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [+375XXXXXXXX@ims_incoming:1] Answer("SIP/ims222-00000001", "") in new stack
       > 0x17dc9a0 -- Probation passed - setting RTP source address to 10.24.0.43:35370
    -- Auto fallthrough, channel 'SIP/ims222-00000001' status is 'UNKNOWN'
[Mar 22 20:46:35] NOTICE[8286][C-00000001]: chan_sip.c:25805 handle_request_invite: Unable to create/find SIP channel for this INVITE

Код: Выделить всё

[ims_incoming]                               
exten => +375XXXXXXXX,1,Answer()             
exten => s,n,Wait(1)