From 338e48e055627d1f999bef275d0fe02d12e6fd68 Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Thu, 19 Mar 2009 16:04:02 +0000 Subject: [PATCH] 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/1.4@183115 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 7215810df9..0e987e4ecf 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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))) -- 2.47.2