[asterisk-commits] tilghman: branch 1.6.0 r234131 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: tilghman
Date: Thu Dec 10 10:30:22 2009
New Revision: 234131

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=234131
Log:
Merged revisions 234129 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r234129 | tilghman | 2009-12-10 10:24:26 -0600 (Thu, 10 Dec 2009) | 16 lines
  
  Merged revisions 234095 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines
    
    When we receive no response at all to our INVITE, allow the channel to be destroyed.
    (closes issue #15627, closes issue #15716, closes issue #16270, closes issue #15356, issue #16382)
     Reported by: falves11, corruptor, dant
     Patches: 
           20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14)
           20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14)
     Tested by: falves11
    Review: https://reviewboard.asterisk.org/r/446/
  ........
................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=234131&r1=234130&r2=234131
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Thu Dec 10 10:30:22 2009
@@ -4523,6 +4523,9 @@
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 			ast_channel_unlock(p->owner);
 		}
+
+		/* Give the channel a chance to act before we proceed with destruction */
+		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 	}
 	sip_pvt_unlock(p);
 	return 0;
@@ -5186,19 +5189,20 @@
 		if (needcancel) {	/* Outgoing call, not up */
 			if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 				/* stop retransmitting an INVITE that has not received a response */
-				struct sip_pkt *cur;
-				for (cur = p->packets; cur; cur = cur->next) {
-					__sip_semi_ack(p, cur->seqno, cur->is_resp, cur->method ? cur->method : find_sip_method(cur->data));
-				}
-
 				/* if we can't send right now, mark it pending */
 				if (p->invitestate == INV_CALLING) {
 					/* We can't send anything in CALLING state */
 					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
-					/* Do we need a timer here if we don't hear from them at all? */
+					__sip_pretend_ack(p);
+					/* Do we need a timer here if we don't hear from them at all? Yes we do or else we will get hung dialogs and those are no fun. */
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 					append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
 				} else {
+					struct sip_pkt *cur;
+
+					for (cur = p->packets; cur; cur = cur->next) {
+						__sip_semi_ack(p, cur->seqno, cur->is_resp, cur->method ? cur->method : find_sip_method(cur->data));
+					}
 					p->invitestate = INV_CANCELLED;
 					/* Send a new request: CANCEL */
 					transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);