[asterisk-commits] branch oej/test-this-branch r33030 - in /team/oej/test-this-branch: ./ build_...

[asterisk-commits at lists.digium.com]

Author: oej
Date: Thu Jun  8 06:46:20 2006
New Revision: 33030

URL: http://svn.digium.com/view/asterisk?rev=33030&view=rev
Log:
Update, fixes. It does compile again.

Modified:
    team/oej/test-this-branch/build_tools/menuselect.c
    team/oej/test-this-branch/channels/chan_sip.c
    team/oej/test-this-branch/rtp.c

Modified: team/oej/test-this-branch/build_tools/menuselect.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/build_tools/menuselect.c?rev=33030&r1=33029&r2=33030&view=diff
==============================================================================
--- team/oej/test-this-branch/build_tools/menuselect.c (original)
+++ team/oej/test-this-branch/build_tools/menuselect.c Thu Jun  8 06:46:20 2006
@@ -24,12 +24,13 @@
  * \brief A menu-driven system for Asterisk module selection
  */
 
-#include "asterisk.h"
 
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
 #include <unistd.h>
+
+#include "asterisk.h"
 
 #include "mxml/mxml.h"
 #include "menuselect.h"

Modified: team/oej/test-this-branch/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/channels/chan_sip.c?rev=33030&r1=33029&r2=33030&view=diff
==============================================================================
--- team/oej/test-this-branch/channels/chan_sip.c (original)
+++ team/oej/test-this-branch/channels/chan_sip.c Thu Jun  8 06:46:20 2006
@@ -4421,7 +4421,6 @@
 		} else if (p->udptl && (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL)) && (sscanf(m, "image %d udptl t38 %n", &x, &len) == 1)) {
 			if (debug)
 				ast_verbose("Got T.38 offer in SDP\n");
-			found = 1;
 			udptlportno = x;
 			
 			if (p->owner && p->lastinvite) {
@@ -4499,7 +4498,7 @@
 				ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
 			} else {
 				/* XXX This could block for a long time, and block the main thread! XXX */
-				hp = ast_gethostbyname(host, &ahp);
+				hp = ast_gethostbyname(host, &audiohp);
 				if (!hp) {
 					ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c);
 				}
@@ -4605,6 +4604,7 @@
 
 	if (udptlportno != -1) {
 		int found = 0;
+		int x;
 
 		old = 0;
 
@@ -8633,7 +8633,7 @@
 			else
 				res = AUTH_SECRET_FAILED;	/* We don't want any guests, fail */
 
-			p->maxcallbitrate = user->maxcallbitrate;
+			p->maxcallbitrate = device->maxcallbitrate;
 			/* If we do not support video, remove video from call structure */
 			if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
 				ast_rtp_destroy(p->vrtp);

Modified: team/oej/test-this-branch/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/rtp.c?rev=33030&r1=33029&r2=33030&view=diff
==============================================================================
--- team/oej/test-this-branch/rtp.c (original)
+++ team/oej/test-this-branch/rtp.c Thu Jun  8 06:46:20 2006
@@ -103,18 +103,10 @@
 /* Forward declarations */
 static int ast_rtcp_write(void *data);
 static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw);
-static int ast_rtcp_send_h261fur(void *data);
 static int ast_rtcp_write_sr(void *data);
 static int ast_rtcp_write_rr(void *data);
 char *ast_rtp_get_quality(struct ast_rtp *rtp);
-static void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local);
 static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);
-
-/*! \brief The value of each payload format mapping: */
-struct rtpPayloadType {
-	int isAstFormat; 	/*!< whether the following code is an AST_FORMAT */
-	int code;
-};
 
 #define FLAG_3389_WARNING		(1 << 0)
 #define FLAG_NAT_ACTIVE			(3 << 1)
@@ -1419,13 +1411,6 @@
 	}
 }
 
-static void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
-	if (rtp)
-		rtp->rtp_offered_from_local = local;
-	else
-		ast_log(LOG_WARNING, "rtp structure is null\n");
-}
-
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
 {
 	struct rtpPayloadType result;
@@ -1433,10 +1418,6 @@
 	result.isAstFormat = result.code = 0;
 	if (pt < 0 || pt > MAX_RTP_PT) 
 		return result; /* bogus payload type */
-
-	/* Start with negotiated codecs */
-	if (!rtp->rtp_offered_from_local)
-		result = rtp->current_RTP_PT[pt];
 
 	/* If it doesn't exist, check our static RTP type list, just in case */
 	if (!result.code)