Merged revisions 141810,141868 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)
  
  
  ........
................
  r141868 | mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4 lines
  
  Um, apparently I didn't actually finish merging before committing.
  Bad bad bad
................


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@187554 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 14 insertions, 3 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6d1949190..59db02f5a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -993,6 +993,7 @@ struct sip_auth {
 
 #define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)	/*!< DP: Compensate for buggy RFC2833 implementations */
 #define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< DP: Buggy CISCO MWI fix */
+#define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 27)       /*!< 29: Has a dialog been established? */
 #define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
 
@@ -5081,6 +5082,7 @@ static int sip_answer(struct ast_channel *ast)
 			change_t38_state(p, T38_ENABLED);
 		}
 		ast_rtp_new_source(p->rtp);
+		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
 	}
 	sip_pvt_unlock(p);
@@ -6138,7 +6140,7 @@ restartsearch:
 			found = (!strcmp(p->callid, callid));
 		else 
 			found = (!strcmp(p->callid, callid) && 
-			(!pedanticsipchecking || ast_strlen_zero(tag) || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ;
+			(!pedanticsipchecking || ast_strlen_zero(tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, tag))) ;
 
 		ast_debug(5, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
 
@@ -15222,6 +15224,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
 
 		/* 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;
@@ -15724,8 +15727,12 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
 				}
 			} 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) 
@@ -17528,9 +17535,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			if (p->t38.state == T38_PEER_REINVITE) {
 				p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, p);
 			} else if (p->t38.state == T38_ENABLED) {
+				ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 				transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)));
 			} else if (p->t38.state == T38_DISABLED) {
 				/* 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); 
 			}
 
@@ -18210,6 +18219,7 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
 		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;
@@ -18483,6 +18493,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
 			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 */
 				ASTOBJ_WRLOCK(p->relatedpeer);
@@ -18499,7 +18510,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
 				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));