From 694d3435e18c9f277b7c29ca415ace5201331663 Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Tue, 30 Sep 2003 23:03:57 +0000
Subject: [PATCH] Add sayunixtime, chan_sip updates for codec negotiation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
apps/Makefile | 2 +-
apps/app_sayunixtime.c | 117 +++++++++++++++++++++++++++++++++++++++++
channels/chan_sip.c | 4 +-
3 files changed, 120 insertions(+), 3 deletions(-)
create mode 100755 apps/app_sayunixtime.c
diff --git a/apps/Makefile b/apps/Makefile
index 16850288f9..0e89a7af6c 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 0000000000..6a6fa62199
--- /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 38af7956f9..d5d59e0625 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);
--
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