From 2099b522d5d625ace1b71be9b0980b73ae538d52 Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Thu, 19 Mar 2009 16:08:47 +0000 Subject: [PATCH] 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index daa439d624..996259f0aa 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4979,11 +4979,6 @@ 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)))