Issue #5937 - Make sure that SIP CANCEL's are retransmitted properly

Importing revision 12495 from 1.2 with changes for svn trunk

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12496 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d09733c5c8b5ce5fdabd0f727e31c00294ba0cf5..56ea9d21dd1c927df8a19622c39f9c5a9dc0e546 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2578,12 +2578,15 @@ static int sip_hangup(struct ast_channel *ast)
        if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
                if (needcancel) {       /* Outgoing call, not up */
                        if (ast_test_flag(p, SIP_OUTGOING)) {
+                               /* stop retransmitting an INVITE that has not received a response */
+                               __sip_pretend_ack(p);
+
+                               /* Send a new request: CANCEL */
                                transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, 0);
                                /* Actually don't destroy us yet, wait for the 487 on our original 
                                   INVITE, but do set an autodestruct just in case we never get it. */
                                ast_clear_flag(&locflags, SIP_NEEDDESTROY);
-                               /* stop retransmitting an INVITE that has not received a response */
-                               __sip_pretend_ack(p);
+
                                sip_scheddestroy(p, 32000);
                                if ( p->initid != -1 ) {
                                        /* channel still up - reverse dec of inUse counter