]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Fix an issue where cancelled outgoing SIP calls would erroneously report the device...
authorMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:04:02 +0000 (16:04 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:04:02 +0000 (16:04 +0000)
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/1.4@183115 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 7215810df9d0c975c7b0ad1bfba11dc3914db92d..0e987e4ecfe2c0a0c8689a9e38dee2df31b4b5cc 100644 (file)
@@ -3670,11 +3670,6 @@ 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)))