[asterisk-commits] mmichelson: branch 1.6.1 r141812 - in /branches/1.6.1: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Sep 8 16:20:01 CDT 2008


Author: mmichelson
Date: Mon Sep  8 16:20:01 2008
New Revision: 141812

URL: http://svn.digium.com/view/asterisk?view=rev&rev=141812
Log:
Merged revisions 141810 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep 2008) | 22 lines

Merged revisions 141809 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines

Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.

(closes issue #11536)
Reported by: ibc
Patches:
      11536v2.patch uploaded by putnopvut (license 60)


........

................

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://svn.digium.com/view/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=141812&r1=141811&r2=141812
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Mon Sep  8 16:20:01 2008
@@ -1035,6 +1035,7 @@
 #define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)	/*!< GP: Should we clean memory from peers after expiry? */
 /* Space for addition of other realtime flags in the future */
 #define SIP_PAGE2_STATECHANGEQUEUE	(1 << 9)	/*!< D: Unsent state pending change exists */
+#define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 29)       /*!< 29: Has a dialog been established? */
 
 #define SIP_PAGE2_VIDEOSUPPORT		(1 << 14)	/*!< DP: Video supported if offered? */
 #define SIP_PAGE2_TEXTSUPPORT		(1 << 15)	/*!< GDP: Global text enable */
@@ -5147,9 +5148,11 @@
 		if (p->t38.state == T38_PEER_DIRECT) {
 			change_t38_state(p, T38_ENABLED);
 			res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+			ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		} else {
 			ast_rtp_new_source(p->rtp);
 			res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
+			ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		}
 	}
 	sip_pvt_unlock(p);
@@ -6230,6 +6233,7 @@
 		}
 	}
 
+<<<<<<< .working
 restartsearch:
 	if (!pedanticsipchecking) {
 		struct sip_pvt tmp_dialog = {
@@ -6249,6 +6253,35 @@
 				ao2_unlock(dialogs);
 				usleep(1);
 				goto restartsearch;
+=======
+	ast_mutex_lock(&iflock);
+	for (p = iflist; p; p = p->next) {
+		/* In pedantic, we do not want packets with bad syntax to be connected to a PVT */
+		int found = FALSE;
+		if (ast_strlen_zero(p->callid))
+			continue;
+		if (req->method == SIP_REGISTER)
+			found = (!strcmp(p->callid, callid));
+		else {
+			found = !strcmp(p->callid, callid);
+			if (pedanticsipchecking && found) {
+				found = ast_strlen_zero(tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, tag);
+			}
+		}
+
+		if (option_debug > 4)
+			ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
+
+		/* If we get a new request within an existing to-tag - check the to tag as well */
+		if (pedanticsipchecking && found  && req->method != SIP_RESPONSE) {	/* SIP Request */
+			if (p->tag[0] == '\0' && totag[0]) {
+				/* We have no to tag, but they have. Wrong dialog */
+				found = FALSE;
+			} else if (totag[0]) {			/* Both have tags, compare them */
+				if (strcmp(totag, p->tag)) {
+					found = FALSE;		/* This is not our packet */
+				}
+>>>>>>> .merge-right.r141809
 			}
 			ao2_unlock(dialogs);
 			return p;
@@ -15354,6 +15387,7 @@
 
 		/* If I understand this right, the branch is different for a non-200 ACK only */
 		p->invitestate = INV_TERMINATED;
+		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
 		check_pendings(p);
 		break;
@@ -15884,8 +15918,12 @@
 				handle_response_notify(p, resp, rest, req, seqno);
 			} else if (sipmethod == SIP_REGISTER) 
 				res = handle_response_register(p, resp, rest, req, seqno);
-			else if (sipmethod == SIP_BYE)		/* Ok, we're ready to go */
+			else if (sipmethod == SIP_BYE) {		/* Ok, we're ready to go */
 				p->needdestroy = 1;
+				ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+			} else if (sipmethod == SIP_SUBSCRIBE) {
+				ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+			}
 			break;
 		case 202:   /* Transfer accepted */
 			if (sipmethod == SIP_REFER) 
@@ -17674,6 +17712,7 @@
 							}
 						} else {
 							/* The other side is already setup for T.38 most likely so we need to acknowledge this too */
+							ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 							transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL);
 							change_t38_state(p, T38_ENABLED);
 						}
@@ -17687,6 +17726,7 @@
 					}
 				} else {
 					/* we are not bridged in a call */
+					ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 					transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL);
 					change_t38_state(p, T38_ENABLED);
 				}
@@ -17713,6 +17753,7 @@
 				/* Respond to normal re-invite */
 				if (sendok) {
 					/* If this is not a re-invite or something to ignore - it's critical */
+					ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 					transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE:TRUE); 
 				}
 			}
@@ -18445,6 +18486,7 @@
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
 	}
+	ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 	transmit_response(p, "200 OK", req);
 
 	return 1;
@@ -18727,6 +18769,7 @@
 			sip_scheddestroy(p, (p->expiry + 10) * 1000);	/* Set timer for destruction of call at expiration */
 
 		if (p->subscribed == MWI_NOTIFICATION) {
+			ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 			transmit_response(p, "200 OK", req);
 			if (p->relatedpeer) {	/* Send first notification */
 				ao2_lock(p->relatedpeer); /* was WRLOCK */
@@ -18743,7 +18786,7 @@
 				p->needdestroy = 1;
 				return 0;
 			}
-
+			ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 			transmit_response(p, "200 OK", req);
 			transmit_state_notify(p, firststate, 1, FALSE);	/* Send first notification */
 			append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));




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