]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Merged revisions 183117 via svnmerge from
authorMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:09:41 +0000 (16:09 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:09:41 +0000 (16:09 +0000)
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
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 5d79080466244b0f6af3e8a9e9a15edb3a110dd9..45d91be8bdf0bdb9127779b59485ff8200812894 100644 (file)
@@ -5235,11 +5235,6 @@ 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)))