[asterisk-bugs] [Asterisk 0016992]: incoming INVITE received no progress, just 200 OK, causing Sipra pstn to go off-hook

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16992 
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Reported By:                jw-asterisk
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16992
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.5 
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:             2010-03-08 22:12 CST
Last Modified:              2010-03-10 18:17 CST
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Summary:                    incoming INVITE received no progress, just 200 OK,
causing Sipra pstn to go off-hook
Description: 
With a Sipura 3102 using asterisk as its proxy (pstn->voip). the Sipura
sends asterisk an INVITE, then asterisk responds with a "100 Trying" almost
immediately,  but then as soon as asterisk starts dialing extensions to
handle the incoming call it sends "200 OK" to the Sipura.  This causes the
Sipura to prematurely go off-hook.

Shouldn't asterisk be sending a "180 Ringing" to the Sipura until one of
the extensions answers the call at which point the channels would be
bridged?

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 (0119246) jw-asterisk (reporter) - 2010-03-10 18:17
 https://issues.asterisk.org/view.php?id=16992#c119246 
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Well there is absolutely no way I am going to be able to do that. Because
asterisk peppers all of its output with a myriad of ^M's (as though it was
running on MS Windows), and it also reveals juicy details like my IP
addresses, port numbers, extension names, and more.

Having recently survived a hackers prolonged attack on my machine which
consumed much of my available bandwidth (you cannot stop UDP packets
arriving on ones doorstep), I am not going to publish handy details about
my configuration.  Sorry.

But, given that SIP worked fine in version 1.2.13, was broken in most
versions 1.4.x that I tried, and is again broken in 1.6.5 one has to wonder
what the version numbers really mean.  They don't seem to imply progress or
improvement.  Why is chan_sip.c being constantly rewritten? Surely it
cannot be that difficult to stabilize the SIP code and have it work
properly in every version. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-10 18:17 jw-asterisk    Note Added: 0119246                          
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