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authorMark Michelson <mmichelson@digium.com>2009-03-19 16:07:54 +0000
committerMark Michelson <mmichelson@digium.com>2009-03-19 16:07:54 +0000
commitabb71e3d55ca5dbb909eb669d996d1e867fb0ee8 (patch)
tree45a90ae49683fccc476f1696c002dfec725366a0 /channels
parent12bd92898de7fd97b5b9437ef7a3e72d29b827a3 (diff)
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
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c5
1 files changed, 0 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b3132ffd9..ef745aecb 100644
--- 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)))