Add sayunixtime, chan_sip updates for codec negotiation

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1589 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 120 insertions, 3 deletions

diff --git a/apps/Makefile b/apps/Makefile
index 16850288f..0e89a7af6 100755
--- a/apps/Makefile
+++ b/apps/Makefile
@@ -24,7 +24,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.
      app_authenticate.so app_softhangup.so app_lookupblacklist.so \
      app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
      app_enumlookup.so app_voicemail2.so app_transfer.so app_setcidnum.so app_cdr.so \
-     app_hasnewvoicemail.so
+     app_hasnewvoicemail.so app_sayunixtime.so
 
 #APPS+=app_sql_postgres.so
 #APPS+=app_sql_odbc.so
diff --git a/apps/app_sayunixtime.c b/apps/app_sayunixtime.c
new file mode 100755
index 000000000..6a6fa6219
--- /dev/null
+++ b/apps/app_sayunixtime.c
@@ -0,0 +1,117 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * SayUnixTime application
+ * 
+ * Copyright (c) 2003 Tilghman Lesher.  All rights reserved.
+ *
+ * Tilghman Lesher <app_sayunixtime__200309@the-tilghman.com>
+ *
+ * This code is in the public domain.
+ *
+ */
+
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/say.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+
+static char *tdesc = "Say time";
+
+static char *app_sayunixtime = "SayUnixTime";
+
+static char *sayunixtime_synopsis = "Says a specified time in a custom format";
+
+static char *sayunixtime_descrip =
+"SayUnixTime([unixtime][|[timezone][|format]])\n"
+"  unixtime: time, in seconds since Jan 1, 1970.  May be negative.\n"
+"              defaults to now.\n"
+"  timezone: timezone, see /usr/share/zoneinfo for a list.\n"
+"              defaults to machine default.\n"
+"  format:   a format the time is to be said in.  See voicemail.conf.\n"
+"              defaults to \"ABdY 'digits/at' IMp\"\n"
+"  Returns 0 or -1 on hangup.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int sayunixtime_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s,*zone=NULL,*timec;
+	time_t unixtime;
+	char *format = "ABdY 'digits/at' IMp";
+	struct timeval tv;
+
+	LOCAL_USER_ADD(u);
+
+	gettimeofday(&tv,NULL);
+	unixtime = (time_t)tv.tv_sec;
+
+	if (data) {
+		s = data;
+		s = strdupa(s);
+		if (s) {
+			timec = strsep(&s,"|");
+			if ((timec) && (*timec != '\0')) {
+				long timein;
+				if (sscanf(timec,"%ld",&timein) == 1) {
+					unixtime = (time_t)timein;
+				}
+			}
+			if (s) {
+				zone = strsep(&s,"|");
+				if (zone && (*zone == '\0'))
+					zone = NULL;
+				if (s) {
+					format = s;
+				}
+			} else {
+				ast_log(LOG_ERROR, "Out of memory error\n");
+			}
+		}
+	}
+
+	res = ast_say_date_with_format(chan, unixtime, AST_DIGIT_ANY, chan->language, format, zone);
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_sayunixtime);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_sayunixtime, sayunixtime_exec, sayunixtime_synopsis, sayunixtime_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 38af7956f..d5d59e062 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2423,7 +2423,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
 	/* Start by sending our preferred codecs */
 	cur = prefs;
 	while(cur) {
-		if (p->capability & cur->codec) {
+		if (p->jointcapability & cur->codec) {
 			if (sipdebug)
 				ast_verbose("Answering with preferred capability %d\n", cur->codec);
 			codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec);
@@ -2445,7 +2445,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
 	}
 	/* Now send any other common codecs, and non-codec formats: */
 	for (x = 1; x <= AST_FORMAT_MAX_AUDIO; x <<= 1) {
-		if ((p->capability & x) && !(alreadysent & x)) {
+		if ((p->jointcapability & x) && !(alreadysent & x)) {
 			if (sipdebug)
 				ast_verbose("Answering with capability %d\n", x);	
 			codec = ast_rtp_lookup_code(p->rtp, 1, x);