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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183117 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b3132ff..ef745ae 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5662,11 +5662,6 @@ static int sip_hangup(struct ast_channel *ast)
                                        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;
                                if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))