[asterisk-bugs] [Asterisk 0018633]: asterisk does not respond with ACK on retransmission of 200 OK after it sent ACK

[Asterisk Bug Tracker]

The following issue has been REOPENED. 
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https://issues.asterisk.org/view.php?id=18633 
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Reported By:                rrevels
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18633
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-01-17 09:31 CST
Last Modified:              2011-02-11 16:00 CST
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Summary:                    asterisk does not respond with ACK on retransmission
of 200 OK after it sent ACK
Description: 
Upstream blocking causes the first ACK to be dropped before hitting
destination.  Retransmitted 200s show up in network trace on Asterisk
server but no ACK response is ever generated
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---------------------------------------------------------------------- 
 (0131882) rrevels (reporter) - 2011-02-11 16:00
 https://issues.asterisk.org/view.php?id=18633#c131882 
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Here is a trace off the network.  I'll get one from the asterisk console
some time next week.

54315  79.405282 12.x.x.x -> 71.x.x.x SIP/SDP Request: INVITE
sip:+19286809743 at 71.x.x.x, with session description
54337  79.433605 71.x.x.x -> 12.x.x.x SIP Status: 100 Giving a try
55598  80.940023 71.x.x.x -> 12.x.x.x SIP/SDP Status: 180 Ringing, with
session description
67894  97.760896 71.x.x.x -> 12.x.x.x SIP/SDP Status: 200 OK, with session
description
67895  97.761117 12.x.x.x -> 71.x.x.x SIP Request: ACK
sip:+19286809743 at 71.x.x.x:9060;transport=udp
68272  98.271138 71.x.x.x -> 12.x.x.x SIP/SDP Status: 200 OK, with session
description
69011  99.271627 71.x.x.x -> 12.x.x.x SIP/SDP Status: 200 OK, with session
description
70513 101.334141 71.x.x.x -> 12.x.x.x SIP/SDP Status: 200 OK, with session
description

The calls are initiated via voice T-1 connected to the asterisk server. 
The call is then forwarded to the VoIP Provider via SIP.  The asterisk
server is on a data T-1.  In the trace above, the asterisk server is the
12.x.x.x IP and the provider is the 71.x.x.x IP.

The dialplan is pretty simple.  

exten => _1X.,1,NoOp(!-- LOGGING - Outbound Call To ${EXTEN} -
${STRFTIME(${EPOCH},GMT-5,%F %T)} --!)
exten => _1X.,n,Set(CALLERID(name)=${siteCNAM})
exten => _1X.,n,Set(CALLERID(num)=${siteCNUM})
exten => _1X.,n,Set(CALLERID(ani)=${siteCNUM})
exten =>
_1X.,n,Dial(SIP/+${EXTEN}@to-provider,,rM(checkSIPid^${CHANNEL}))
exten => _1X.,n,Goto(result-${DIALSTATUS},${EXTEN},1) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-11 16:00 rrevels        Note Added: 0131882                          
2011-02-11 16:00 rrevels        Status                   closed => new       
2011-02-11 16:00 rrevels        Resolution               suspended => reopened
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