[svn-commits] branch oej/strictrouting r36778 - in /team/oej/strictrouting: ./ channels/

svn-commits at lists.digium.com svn-commits at lists.digium.com
Sun Jul 2 23:55:54 MST 2006


Author: oej
Date: Mon Jul  3 01:55:53 2006
New Revision: 36778

URL: http://svn.digium.com/view/asterisk?rev=36778&view=rev
Log:
Reset automerge

Modified:
    team/oej/strictrouting/   (props changed)
    team/oej/strictrouting/channels/chan_sip.c

Propchange: team/oej/strictrouting/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Propchange: team/oej/strictrouting/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jul  3 01:55:53 2006
@@ -1,1 +1,1 @@
-/trunk:1-36636
+/trunk:1-36696

Modified: team/oej/strictrouting/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/strictrouting/channels/chan_sip.c?rev=36778&r1=36777&r2=36778&view=diff
==============================================================================
--- team/oej/strictrouting/channels/chan_sip.c (original)
+++ team/oej/strictrouting/channels/chan_sip.c Mon Jul  3 01:55:53 2006
@@ -5153,7 +5153,7 @@
 /*! \brief Add route header into request per learned route */
 static void add_route(struct sip_request *req, struct sip_route *route)
 {
-	char r[256], *p;
+	char r[BUFSIZ], *p;
 	int n, rem = sizeof(r);
 
 	if (!route)
@@ -5317,7 +5317,7 @@
 	return 0;
 }
 
-/*! \brief Initialize a SIP request packet within an existing dialog */
+/*! \brief Initialize a SIP request message (not the initial one in a dialog) */
 static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
 {
 	struct sip_request *orig = &p->initreq;
@@ -5351,20 +5351,20 @@
 		is_strict = TRUE;
 	}
 
-	if (sipmethod == SIP_CANCEL) {		/* Cancel goes to the INVITEs URI */
+	if (sipmethod == SIP_CANCEL)
 		c = p->initreq.rlPart2;	/* Use original URI */
-	} else if (sipmethod == SIP_ACK) {	/* SIP_ACK has two routing scenarios */
+	else if (sipmethod == SIP_ACK) {
 		/* Use URI from Contact: in 200 OK (if INVITE) 
 		(we only have the contacturi on INVITEs) */
 		if (!ast_strlen_zero(p->okcontacturi))
 			c = is_strict ? p->route->hop : p->okcontacturi;
  		else
  			c = p->initreq.rlPart2;
-	} else if (!ast_strlen_zero(p->okcontacturi)) {
+	} else if (!ast_strlen_zero(p->okcontacturi)) 
 		c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
-	} else if (!ast_strlen_zero(p->uri)) {
+	else if (!ast_strlen_zero(p->uri)) 
 		c = p->uri;
-	} else {
+	else {
 		char *n;
 		/* We have no URI, use To: or From:  header as URI (depending on direction) */
 		ast_copy_string(stripped, get_header(orig, (ast_test_flag(&p->flags[0], SIP_OUTGOING)) ? "To" : "From"),
@@ -7576,7 +7576,8 @@
 			if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
 				/* ast_calloc is not needed because all fields are initialized in this block */
 				ast_copy_string(thishop->hop, rr, len);
-				ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
+				if (option_debug > 1)
+					ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop);
 				/* Link in */
 				if (backwards) {
 					/* Link in at head so they end up in reverse order */
@@ -7605,9 +7606,8 @@
 		/* Can be multiple Contact headers, comma separated values - we just take the first */
 		contact = get_header(req, "Contact");
 		if (!ast_strlen_zero(contact)) {
-			if (option_debug)
+			if (option_debug > 1)
 				ast_log(LOG_DEBUG, "build_route: Contact hop: %s\n", contact);
-
 			/* Look for <: delimited address */
 			c = get_in_brackets(contact);
 
@@ -12950,7 +12950,8 @@
 			break;
 		case AST_STATE_UP:
 			if (option_debug > 1)
-				ast_log(LOG_DEBUG, "%s: New call is UP.... \n", c->name);
+				ast_log(LOG_DEBUG, "%s: This call is UP.... \n", c->name);
+
 			if (p->t38.state == T38_PEER_REINVITE) {
 				struct ast_channel *bridgepeer = NULL;
 				struct sip_pvt *bridgepvt = NULL;
@@ -12997,6 +12998,8 @@
 						ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
 				}
 			} else if (p->t38.state == T38_DISABLED) { /* Channel doesn't have T38 offered or enabled */
+				int sendok = TRUE;
+
 				/* If we are bridged to a channel that has T38 enabled than this is a case of RTP re-invite after T38 session */
 				/* so handle it here (re-invite other party to RTP) */
 				struct ast_channel *bridgepeer = NULL;
@@ -13004,6 +13007,7 @@
 				if ((bridgepeer = ast_bridged_channel(p->owner))) {
 					if (!strcasecmp(bridgepeer->tech->type, sip_tech.type)) {
 						bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
+						/* Does the bridged peer have T38 ? */
 						if (bridgepvt->t38.state == T38_ENABLED) {
 							ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
 							/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
@@ -13012,13 +13016,14 @@
 							else
 								transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
 							ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
-						} else {
-							/* No bridged peer with T38 enabled*/
-							transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
-						}
+							sendok = FALSE;
+						} 
+						/* No bridged peer with T38 enabled*/
 					}
-				} else	/* No bridged peer */
+				} 
+				if (sendok)
 					transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
+
 			}
 			break;
 		default:



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