[asterisk-bugs] [Asterisk 0018081]: SIP 200 OK withou ACK

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18081 
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Reported By:                marhbere
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18081
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 262803 
Request Review:              
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Date Submitted:             2010-09-30 16:35 CDT
Last Modified:              2010-10-06 12:14 CDT
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Summary:                    SIP  200 OK  withou ACK
Description: 
We have the issue when happening a problem into network.

When Receive INVITE and we respond 200 OK, and this last not arrived to
the originator, then the originator resend INVITE many times. But the
*Asterisk "Answers" all this call such as if it were differents calls.

I guess this is heavy problem when will try conciliation with CDR calls.

Please can tell me if we have any mistake

Tanks
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 (0127760) marhbere (reporter) - 2010-10-06 12:14
 https://issues.asterisk.org/view.php?id=18081#c127760 
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@schmidts, I have tried with pedntic=yes, but still happening the same.
The call is logged such as if this have been two calls; and with
billsec=3.
I know isn´t often that happen, but in a enviroment production it is to
happended to us, And I consider is a conceptual case of how to asterisk
must manage the Flow sip calls.

sip show settings 


Global Settings:
----------------
  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           Disabled
  TLS SIP Port:           Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    No
  Allow overlap dialing:  No
  Allow promsic. redir:   No
  Enable call counters:   Yes
  SIP domain support:     Yes
  Realm. auth:            No
  Our auth realm          srv.teleCompany.com
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    No
  Direct RTP setup:       No
  User Agent:             MyCompanyIVRv1
  SDP Session Name:       Asterisk PBX SVN-branch-1.6.2-r281912
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Call Events:            On
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms

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
  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:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  30 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  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:       200
  Timer B:                32000
  No premature media:     No

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                default
  Nat:                    RFC3581
  DTMF:                   auto
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        No
  Language:               en
  MOH Interpret:          default
  MOH Suggest:            default
  Voice Mail Extension:   asterisk
  Forward Detected Loops: Yes

Thanks,
Marcelo B. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-06 12:14 marhbere       Note Added: 0127760                          
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