История изменений [назад]

Asterisk не принимает часть SIP пакетов. В tcpdump их видно, а в асериске с включеным дебагом - нет. Иногда не принимает Invite, иногда Bye.

Asterisk не принимает часть SIP пакетов. В tcpdump их видно, а в асериске с включеным дебагом - нет. Иногда не принимает Invite, иногда Bye.

Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           Yes
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  Always auth rejects:    Yes
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              Unknown
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  T38 fax pt UDPTL:       No
  RFC2833 Compensation:   No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 0x28000c (ulaw|alaw|h263|h264)
  Codec Order:            ulaw:20,alaw:20
  T1 minimum:             100
  Relax DTMF:             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

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

Asterisk не принимает часть SIP пакетов. В tcpdump их видно, а в асериске с включеным дебагом - нет. Иногда не принимает Invite, иногда Bye.

Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           Yes
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  Always auth rejects:    Yes
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              Unknown
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  T38 fax pt UDPTL:       No
  RFC2833 Compensation:   No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 0x28000c (ulaw|alaw|h263|h264)
  Codec Order:            ulaw:20,alaw:20
  T1 minimum:             100
  Relax DTMF:             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

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