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geisi

Registriert seit: 19. Sep 2003
449 Beiträge
 
Delphi 6 Professional
 
#6

Re: Asterisk legt nicht auf, wenn caller auflegt

  Alt 30. Mär 2010, 07:25
hallo, sieht vielleicht jemand einen fehler in meinen sip einstellungen?
Code:
Global Settings:
----------------
  UDP SIP Port:          5060
  UDP Bindaddress:       0.0.0.0
  TCP SIP Port:          Disabled
  TLS SIP Port:          Disabled
  Videosupport:          No
  Textsupport:           No
  AutoCreate Peer:       No
  Ignore SDP sess. ver.: No
  Match Auth Username:   No
  Allow unknown access:  Yes
  Allow subscriptions:   Yes
  Allow overlap dialing: Yes
  Allow promsic. redir:  No
  Enable call counters:  No
  SIP domain support:    No
  Realm. auth:           No
  Our auth realm         asterisk
  Call to non-local dom.: Yes
  URI user is phone no:  No
  Always auth rejects:   No
  Call limit peers only: Yes
  Direct RTP setup:      No
  User Agent:            Asterisk PBX 1.6.0.17
  SDP Session Name:      Asterisk PBX 1.6.0.17
  SDP Owner Name:        root
  Reg. context:          (not set)
  Regexten on Qualify:   No
  Caller ID:             Unknown
  From: Domain:          
  Record SIP history:    Off
  Call Events:           Off
  T38 fax pt UDPTL:      No
  SIP realtime:          Disabled
  Qualify Freq :         60000 ms

Network QoS Settings:
---------------------------
  IP ToS SIP:            CS3
  IP ToS RTP audio:      EF
  IP ToS RTP video:      AF41
  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
  Jitterbuffer forced:   No
  Jitterbuffer max size: -1
  Jitterbuffer resync:   -1
  Jitterbuffer impl:    
  Jitterbuffer log:      No

Network Settings:
---------------------------
  SIP address remapping: Disabled, no localnet list
  Externhost:            <none>
  Externip:              0.0.0.0:0
  Externrefresh:         10
  Internal IP:           127.0.0.1:5060
  STUN server:           0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                0xc (ulaw|alaw)
  Codec Order:           ulaw:20,alaw:20
  Relax DTMF:            No
  RFC2833 Compensation:  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:  No
  Reg. min duration      60 secs
  Reg. max duration:     3600 secs
  Reg. default duration: 120 secs
  Outbound reg. timeout: 20 secs
  Outbound reg. attempts: 0
  Notify ringing state:  Yes
  Notify hold state:     Yes
  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

Default Settings:
-----------------
  Context:               from-sip-external
  Nat:                   RFC3581
  DTMF:                  rfc2833
  Qualify:               0
  Use ClientCode:        No
  Progress inband:       Never
  Language:              
  MOH Interpret:         default
  MOH Suggest:          
  Voice Mail Extension:  *97

----
mfg geisi
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