Index: asterisk-1.4.speakup.26.1/channels/chan_sip.c
===================================================================
--- asterisk-1.4.speakup.26.1.orig/channels/chan_sip.c	2009-08-13 11:40:28.000000000 +0200
+++ asterisk-1.4.speakup.26.1/channels/chan_sip.c	2009-10-05 11:51:13.000000000 +0200
@@ -948,6 +948,7 @@
 		AST_STRING_FIELD(peermd5secret);
 		AST_STRING_FIELD(cid_num);	/*!< Caller*ID number */
 		AST_STRING_FIELD(cid_name);	/*!< Caller*ID name */
+		AST_STRING_FIELD(via_branch);	/*!< branch paramater of first Via header */
 		AST_STRING_FIELD(fullcontact);	/*!< The Contact: that the UA registers with us */
 		AST_STRING_FIELD(our_contact);	/*!< Our contact header */
 		AST_STRING_FIELD(rpid);		/*!< Our RPID header */
@@ -4691,18 +4692,58 @@
 	return p;
 }
 
+/*! \brief check Via: header for branch paramater */
+static char *get_branch(const struct sip_request *req, char *branch)
+{
+	char via[512];
+	char *c, *b;
+
+	ast_copy_string(via, get_header(req, "Via"), sizeof(via));
+
+	/* Work on the leftmost value of the topmost Via header */
+	c = strchr(via, ',');
+	if (c)
+		*c = '\0';
+
+	if (option_debug)
+		ast_log(LOG_DEBUG, "First Via: %s\n", via);
+
+	/* Check for branch */
+	b = strstr(via, ";branch=");
+	if (!b) {
+		ast_log(LOG_DEBUG, "No branch in Via\n");
+		*branch = '\0';
+		return branch;
+	}
+	b = b + 8;
+
+	c = strchr(b, ';');
+	if (c)
+		*c = '\0';
+
+	ast_copy_string(branch, b, 512);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "This is branch: %s\n", branch);
+
+	return branch;
+}
+
 /*! \brief Connect incoming SIP message to current dialog or create new dialog structure
 	Called by handle_request, sipsock_read */
 static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
 {
 	struct sip_pvt *p = NULL;
-	char *tag = "";	/* note, tag is never NULL */
 	char totag[128];
 	char fromtag[128];
 	const char *callid = get_header(req, "Call-ID");
 	const char *from = get_header(req, "From");
 	const char *to = get_header(req, "To");
 	const char *cseq = get_header(req, "Cseq");
+	int seqno;
+	char *our_fromtag;
+	char *our_totag;
+	char branch[512];
+	int found = FALSE;
 
 	/* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
 	/* get_header always returns non-NULL so we must use ast_strlen_zero() */
@@ -4714,56 +4755,116 @@
 		/* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
 		   we need more to identify a branch - so we have to check branch, from
 		   and to tags to identify a call leg.
-		   For Asterisk to behave correctly, you need to turn on pedanticsipchecking
+		   For Asterisk to behave almost correctly, you need to turn on pedanticsipchecking
 		   in sip.conf
 		   */
 		if (gettag(req, "To", totag, sizeof(totag)))
 			ast_set_flag(req, SIP_PKT_WITH_TOTAG);	/* Used in handle_request/response */
 		gettag(req, "From", fromtag, sizeof(fromtag));
 
-		tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
-
-		if (option_debug > 4 )
-			ast_log(LOG_DEBUG, "= Looking for  Call ID: %s (Checking %s) --From tag %s --To-tag %s  \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);
+		if (option_debug > 4)
+			ast_log(LOG_DEBUG, "= Looking for Call-ID: %s   From-tag %s   To-tag %s  \n", callid, fromtag, totag);
 	}
 
 	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))
+		/* try and find a channel with a matching callid */
+		if (ast_strlen_zero(p->callid) || strcmp(p->callid, 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);
+			ast_log(LOG_DEBUG, "Considering channel: %p tag: %s theirtag: %s state: %d uri: %s contact: %s\n", p, p->tag, p->theirtag, p->invitestate, p->uri, p->our_contact);
 
-		/* 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 */
-				}
+		if (!pedanticsipchecking || req->method == SIP_REGISTER) {
+			found = TRUE;
+			break;
+		}
+
+		/* tags depend on the direction of the request */
+		if (req->method == SIP_RESPONSE) {
+			our_fromtag = p->tag;
+			our_totag = p->theirtag;
+		} else {
+			our_fromtag = p->theirtag;
+			our_totag = p->tag;
+		}
+
+		/* from-tags must ALWAYS match */
+		if (strcmp(our_fromtag, fromtag)) {
+			if (option_debug > 4)
+				ast_log(LOG_DEBUG, "From-tags do not match. local tag: %s remote tag: %s\n", our_fromtag, fromtag);
+			continue;
+		}
+
+		/* Check initial INVITE */
+		if ((req->method == SIP_INVITE) && ast_strlen_zero(totag)) {
+			/* Check for retransmission
+			 * CSeq and branch must be identical.
+			 * Skip CSeq check as we don't keep a history of CSeq's in the pvt :( */
+
+			get_branch(req, branch);
+			if (strcmp(branch, p->via_branch) == 0) {
+				if (option_debug > 4)
+					ast_log(LOG_DEBUG, "Match: Retransmission\n");
+				found = TRUE;
+				break;
+			}
+
+			/* If the dialog is already up, absorb the INVITE
+			 * as it may be a late fork and we don't want 2 dialogs */
+			if (ast_test_flag(&(p->flags[1]), SIP_PAGE2_DIALOG_ESTABLISHED)) {
+				if (option_debug > 4)
+					ast_log(LOG_DEBUG, "Match: Late initial INVITE for established dialog\n");
+				found = TRUE;
+				break;
+			}
+
+			/* We must match when authentication is used (CSEQ = previous CSEQ + 1) */
+			sscanf(get_header(req, "CSeq"), "%d ", &seqno);
+			if (option_debug > 4)
+				ast_log(LOG_DEBUG, "CSeq: %s  seqno: %d  icseq: %d\n", cseq, seqno, p->icseq);
+			if (seqno == (p->icseq + 1)) {
+				if (option_debug > 4)
+					ast_log(LOG_DEBUG, "Match: Authentication attempt for previous request\n");
+				found = TRUE;
+				break;
 			}
-			if (!found && option_debug > 4)
-				ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text);
+
+			/* Every other initial INVITE must be a forked request and will be treated as new dialog */
+			continue;
 		}
-		if (found) {
-			/* Found the call */
-			ast_mutex_lock(&p->lock);
-			ast_mutex_unlock(&iflock);
-			return p;
+
+		/* Check to-tags */
+		if (strcmp(our_totag, totag) == 0) {
+			/* to-tags match */
+			if (option_debug > 4)
+				ast_log(LOG_DEBUG, "Match: both tags matched\n");
+			found = TRUE;
+			break;
+		} else {
+			/* to-tags must match for in-dialog requests and ACK's*/
+			if (ast_test_flag(&(p->flags[1]), SIP_PAGE2_DIALOG_ESTABLISHED)
+				|| ((p->invitestate >= INV_COMPLETED) && p->invitestate != INV_CANCELLED)) {
+				/* to-tags differ */
+				if (option_debug > 4)
+					ast_log(LOG_DEBUG, "To-tags do not match for in-dialog request. local tag: %s remote tag: %s\n", our_totag, totag);
+				continue;
+			}
 		}
+
+		/* There is not an established dialog for this pvt. Allow any to-tag */
+		if (option_debug > 4)
+			ast_log(LOG_DEBUG, "Match: to-tags ignored in early-dialog\n");
+		found = TRUE;
+		break;
+	}
+
+	if (found) {
+		/* Found the call */
+		ast_log(LOG_DEBUG, "%s Found channel: %p, local tag: %s remote tag: %s\n", sip_methods[req->method].text , p, p->tag, p->theirtag);
+		ast_mutex_lock(&p->lock);
+		ast_mutex_unlock(&iflock);
+		return p;
 	}
 	ast_mutex_unlock(&iflock);
 
@@ -11519,6 +11620,7 @@
 			ast_cli(fd, "  Audio IP:               %s %s\n", ast_inet_ntoa(cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" );
 			ast_cli(fd, "  Our Tag:                %s\n", cur->tag);
 			ast_cli(fd, "  Their Tag:              %s\n", cur->theirtag);
+			ast_cli(fd, "  Branch:                 %s\n", cur->via_branch);
 			ast_cli(fd, "  SIP User agent:         %s\n", cur->useragent);
 			if (!ast_strlen_zero(cur->username))
 				ast_cli(fd, "  Username:               %s\n", cur->username);
@@ -13260,7 +13362,6 @@
 				res = handle_response_register(p, resp, rest, req, seqno);
 			else if (sipmethod == SIP_BYE) {		/* Ok, we're ready to go */
 				ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
-				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;
@@ -14500,10 +14601,15 @@
 	const char *p_replaces;
 	char *replace_id = NULL;
 	const char *required;
+	char branch[512];
 	unsigned int required_profile = 0;
 	struct ast_channel *c = NULL;		/* New channel */
 	int reinvite = 0;
 
+	/* Remember the branch from the Via header */
+	get_branch(req, branch);
+	ast_string_field_set(p, via_branch, branch);
+
 	/* Find out what they support */
 	if (!p->sipoptions) {
 		const char *supported = get_header(req, "Supported");
@@ -15720,7 +15826,6 @@
 		if (option_debug > 2)
 			ast_log(LOG_DEBUG, "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;