[asterisk-commits] mmichelson: branch 1.4 r183115 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Thu Mar 19 11:04:02 2009
New Revision: 183115

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183115
Log:
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/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=183115&r1=183114&r2=183115
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Mar 19 11:04:02 2009
@@ -3669,11 +3669,6 @@
 					   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;