[asterisk-bugs] [Asterisk 0017071]: When using another SIP Trunk, Asterisk generates the initial ring RING as a response to "SIP SESSION PROGRESS"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17071 
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Reported By:                Alex Oniciuc
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17071
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                       SWP-1137 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-22 05:43 CDT
Last Modified:              2010-03-24 03:18 CDT
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Summary:                    When using another SIP Trunk, Asterisk generates the
initial ring RING as a response to "SIP SESSION PROGRESS"
Description: 
I’m having a strange problem with the VoIP Gateway that I’m using to go
on the PSTN: if the number on the other end is busy or unavailable I hear
an initial RING, generated by Asterisk from what I’m seeing and after
that the line goes down with busy signal:

Here is the scenario:

    Softphone    *ASTERISK                PATTON                   PSTN
[BUSY]

1   INVITE     >  INVITE              >   INVITE
2.	                              <   SIP/2.0 100 Trying
3.  RING      <  SIP/2.0 180 Ringing  <   SIP/2.0 183 Session Progress
4.	      <  SIP/2.0 603 Declined <   SIP/2.0 406 Not Acceptable

Is this regular? Asterisk isn’t supposed to generate the RING  only
after the first one received from the PATTON?

This can be very annoying because the calling party may have the
impression that the remote party hang up.
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 (0119788) Alex Oniciuc (reporter) - 2010-03-24 03:18
 https://issues.asterisk.org/view.php?id=17071#c119788 
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OK, I've commented this part
<snip>

 /*
else {
     Alcatel PBXs are known to send 183s with no SDP after sending
     * a 100 Trying response. We're just going to treat this sort of
thing
     * the same as we would treat a 180 Ringing

     if (!req->ignore && p->owner) {
         ast_queue_control(p->owner, AST_CONTROL_RINGING);
     }
}
*/
</snip>

and now it's working like a charm.

The developer that implemented this didn't consider that the Alcatel PBX
could return a Busy/Unreachable state after SIP 183.

It would be nice to have this part removed from the official release
though. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-24 03:18 Alex Oniciuc   Note Added: 0119788                          
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