[asterisk-commits] mmichelson: branch 1.6.0 r183118 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Thu Mar 19 11:08:47 2009
New Revision: 183118

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183118
Log:
Merged revisions 183117 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines
  
  Merged revisions 183115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
    
    Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
    
    A user was having an issue where if an outgoing SIP call was canceled, the SIP device
    would remain in use if we had not received any response to the initial INVITE we sent out.
    The SIP device would remain in use until the autocongestion timer was exhausted.
    
    I tracked down the cause of this to be the section of code I am removing here. I asked several
    people what the purpose of this code was meant to be, but no one could give me any sort of
    answer as to why this was here. The person who was having this issue has been using this patch
    for several months and it has stopped the problems they have had.
    
    AST-196
  ........
................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=183118&r1=183117&r2=183118
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Thu Mar 19 11:08:47 2009
@@ -4978,11 +4978,6 @@
 					   INVITE, but do set an autodestruct just in case we never get it. */
 					needdestroy = 0;
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-				}
-				if ( p->initid != -1 ) {
-					/* channel still up - reverse dec of inUse counter
-					   only if the channel is not auto-congested */
-					update_call_counter(p, INC_CALL_LIMIT);
 				}
 			} else {	/* Incoming call, not up */
 				const char *res;