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

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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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:                     closed
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:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2010-09-30 16:35 CDT
Last Modified:              2010-10-06 15:44 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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 (0127767) schmidts (manager) - 2010-10-06 15:44
 https://issues.asterisk.org/view.php?id=18081#c127767 
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i close this issue cause i have seen there is nothing wrong after catching
what i have missed before.
asterisk really act like it should act in this case. you see the response
to the first invite which will be retransmitted and on every further invite
there is only one (unreliable) responde which is exactly the same like the
first respond. This is just the way RFC said what should happens also if
the Call is allready UP.
Asterisk does not start a new call, its still only one call. 

how long is the we-dont-have-tech-support file you use for playback? if i
have to guess i would say exactly 3 seconds. 
Cause in your log you can see this row:
[Oct  5 21:36:26] VERBOSE[10748] pbx.c:     -- Auto fallthrough, channel
'SIP/EXT-SRVARG009-00002775' status is 'UNKNOWN'
After looking into the main pbx function i see why this happens, the call
is still in an undefinate state cause there was no ACK to the 200 OK
message. 
Thats why the pbx function breaks further running through the dialplan
after 3 seconds (as i guess after the playback is finished) and makes a CDR
Update to save the information.
This cdr information means nothing else than the call is answered (200 ok
was sent so its answered) and until now 3 seconds passed by. You see this
information in your cdrs.
And further this is exactly what asterisk should do in this case cause
reseting the cdr informations could produce massive side effects (think of
an reinvite message after a one hour call which could lead to the same
result)

So i would suggest to use the Answer application before the playback so
playback would not even start cause the channel is not really up or another
way could be to set a channel variable and create a h extension to do a
CDRRESET if this channel var is empty.

best regards

Stefan Schmidt 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-06 15:44 schmidts       Note Added: 0127767                          
2010-10-06 15:44 schmidts       Status                   feedback => closed  
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