[asterisk-commits] dvossel: branch 1.6.1 r248399 - in /branches/1.6.1: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Feb 23 10:48:42 CST 2010


Author: dvossel
Date: Tue Feb 23 10:48:37 2010
New Revision: 248399

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=248399
Log:
Merged revisions 248397 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines
  
  Merged revisions 248396 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines
    
    fixes invite with replaces deadlock
    
    (closes issue #16862)
    Reported by: pwalker
    Patches:
          replaces_deadlock_1.4 uploaded by dvossel (license 671)
    Tested by: pwalker, dvossel
  ........
................

Modified:
    branches/1.6.1/   (props changed)
    branches/1.6.1/channels/chan_sip.c

Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=248399&r1=248398&r2=248399
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Tue Feb 23 10:48:37 2010
@@ -18757,6 +18757,7 @@
 	int gotdest;
 	const char *p_replaces;
 	char *replace_id = NULL;
+	int refer_locked = 0;
 	const char *required;
 	unsigned int required_profile = 0;
 	struct ast_channel *c = NULL;		/* New channel */
@@ -18791,7 +18792,8 @@
 			p->invitestate = INV_COMPLETED;
 			if (!p->lastinvite)
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-			return -1;
+			res = -1;
+			goto request_invite_cleanup;
 		}
 	}
 
@@ -18818,7 +18820,8 @@
 			transmit_response(p, "482 Loop Detected", req);
 			p->invitestate = INV_COMPLETED;
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		} else {
 			/*! This is a spiral. What we need to do is to just change the outgoing INVITE
 			 * so that it now routes to the new Request URI. Since we created the INVITE ourselves
@@ -18844,7 +18847,8 @@
 			 */
 			ast_string_field_set(p->owner, call_forward, peerorhost);
 			ast_queue_control(p->owner, AST_CONTROL_BUSY);
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		}
 	}
 
@@ -18882,7 +18886,8 @@
 			transmit_response_reliable(p, "491 Request Pending", req);
 			ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
 			/* Don't destroy dialog here */
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		}
 	}
 
@@ -18899,7 +18904,8 @@
 			ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
 			transmit_response_reliable(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
 			/* Do not destroy existing call */
-			return -1;
+			res = -1;
+			goto request_invite_cleanup;
 		}
 
 		if (sipdebug)
@@ -18913,7 +18919,8 @@
 			append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			p->invitestate = INV_COMPLETED;
-			return -1;
+			res = -1;
+			goto request_invite_cleanup;
 		}
 
 		/*  Todo: (When we find phones that support this)
@@ -18949,6 +18956,8 @@
 			ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
 			transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
 			error = 1;
+		} else {
+			refer_locked = 1;
 		}
 
 		/* At this point, bot the pvt and the owner of the call to be replaced is locked */
@@ -18988,8 +18997,10 @@
 					ast_channel_unlock(p->refer->refer_call->owner);
 				}
 			}
+			refer_locked = 0;
 			p->invitestate = INV_COMPLETED;
-			return -1;
+			res = -1;
+			goto request_invite_cleanup;
 		}
 	}
 
@@ -19022,7 +19033,8 @@
 					transmit_response_reliable(p, "488 Not acceptable here", req);
 					if (!p->lastinvite)
 						sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-					return -1;
+					res = -1;
+					goto request_invite_cleanup;
 				}
 				ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
 			} else {
@@ -19042,7 +19054,8 @@
 		res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
 		if (res == AUTH_CHALLENGE_SENT) {
 			p->invitestate = INV_COMPLETED;		/* Needs to restart in another INVITE transaction */
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		}
 		if (res < 0) { /* Something failed in authentication */
 			if (res == AUTH_FAKE_AUTH) {
@@ -19055,7 +19068,8 @@
 			p->invitestate = INV_COMPLETED;	
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			ast_string_field_set(p, theirtag, NULL);
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		}
 
 		/* If T38 is needed but not present, then make it magically appear */
@@ -19072,7 +19086,8 @@
 				p->invitestate = INV_COMPLETED;	
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				ast_debug(1, "No compatible codecs for this SIP call.\n");
-				return -1;
+				res = -1;
+				goto request_invite_cleanup;
 			}
 			if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
 				if (p->rtp) {
@@ -19107,7 +19122,8 @@
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				p->invitestate = INV_COMPLETED;	
 			}
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		}
 		gotdest = get_destination(p, NULL);	/* Get destination right away */
 		get_rdnis(p, NULL);			/* Get redirect information */
@@ -19134,7 +19150,8 @@
 			p->invitestate = INV_COMPLETED;	
 			update_call_counter(p, DEC_CALL_LIMIT);
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-			return 0;
+			res = 0;
+			goto request_invite_cleanup;
 		} else {
 
 			/* If no extension was specified, use the s one */
@@ -19188,7 +19205,8 @@
 				if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				}
-				return -1;
+				res = -1;
+				goto request_invite_cleanup;
 			}
 		}
 
@@ -19202,7 +19220,8 @@
        	   			if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				}
-				return -1;
+				res = -1;
+				goto request_invite_cleanup;
 			}
 		}
 
@@ -19216,7 +19235,8 @@
 				if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				}
-				return -1;
+				res = -1;
+				goto request_invite_cleanup;
 			}
 
 			p->stimer->st_active_peer_ua = TRUE;
@@ -19246,7 +19266,8 @@
 				if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				}
-				return -1;
+				res = -1;
+				goto request_invite_cleanup;
 			}
 			break;
 
@@ -19306,7 +19327,10 @@
 		/* Go and take over the target call */
 		if (sipdebug)
 			ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
-		return handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+		res = handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+		refer_locked = 0;
+		goto request_invite_cleanup;
+
 	}
 
 
@@ -19430,6 +19454,16 @@
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		}
 	}
+
+request_invite_cleanup:
+
+	if (refer_locked && p->refer && p->refer->refer_call) {
+		sip_pvt_unlock(p->refer->refer_call);
+		if (p->refer->refer_call->owner) {
+			ast_channel_unlock(p->refer->refer_call->owner);
+		}
+	}
+
 	return res;
 }
 




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