Index: channels/chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.170
diff -u -r1.170 chan_sip.c
--- channels/chan_sip.c	25 Aug 2003 14:17:14 -0000	1.170
+++ channels/chan_sip.c	29 Aug 2003 21:06:23 -0000
@@ -2803,6 +2803,31 @@
 	return send_request(p, &req, 1, p->ocseq);
 }
 
+static int transmit_reset(struct sip_pvt *p)
+{
+	struct sip_request req;
+	char tmp[256];
+	char tmp2[256];
+	char clen[20];
+	initreqprep(&req, p, "NOTIFY", NULL);
+	add_header(&req, "Event", "check-sync");
+	add_header(&req, "Content-Type", notifymime);
+
+	snprintf(clen, sizeof(clen), "%d", strlen(tmp) + strlen(tmp2));
+	add_header(&req, "Content-Length", clen);
+	add_line(&req, tmp);
+	add_line(&req, tmp2);
+
+	if (!p->initreq.headers) {
+		/* Use this as the basis */
+		copy_request(&p->initreq, &req);
+		parse(&p->initreq);
+		determine_firstline_parts(&p->initreq);
+	}
+
+	return send_request(p, &req, 1, p->ocseq);
+}
+
 static int transmit_register(struct sip_registry *r, char *cmd, char *auth, char *authheader);
 
 static int sip_reregister(void *data) 
@@ -3975,6 +4000,66 @@
 #undef FORMAT2
 }
 
+static int sip_send_reset_to_peer(struct sip_peer *peer)
+{
+	/* Derived from sip_send_mwi_to_peer */
+	/* Called with peerl lock, but releases it */
+	
+	struct sip_pvt *p;
+	char name[256] = "";
+	
+	p = sip_alloc(NULL, NULL, 0);
+	if (!p) {
+		ast_log(LOG_WARNING, "Unable to build sip pvt data for RESET\n");
+		ast_mutex_unlock(&peerl.lock);
+		return -1;
+	}
+	strncpy(name, peer->name, sizeof(name) - 1);
+	ast_mutex_unlock(&peerl.lock);
+	if (create_addr(p, name)) {
+		/* Maybe they're not registered, etc. */
+		sip_destroy(p);
+		return 0;
+	}
+	/* Recalculate our side, and recalculate Call ID */
+	if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip))
+		memcpy(&p->ourip, &__ourip, sizeof(p->ourip));
+	/* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
+	snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x", inet_ntoa(p->ourip), ourport, p->branch);
+	build_callid(p->callid, sizeof(p->callid), p->ourip);
+	
+	transmit_reset(p);
+		
+	sip_scheddestroy(p, 15000);
+	return 0;
+}
+
+static int sip_reset_phone_peer(int fd, int argc, char *argv[])
+{
+	struct sip_peer *peer;
+	if (argc != 4)
+		return RESULT_SHOWUSAGE;	
+	ast_mutex_lock(&peerl.lock);
+	
+	/* Check incoming extension request.  If it is 'all' then reset all SIP phones.
+	   Otherwise, just reset the specified extension */
+	
+	for (peer = peerl.peers;peer;peer = peer->next) {
+		if (!strcmp(argv[3],"all")) {
+				ast_log(LOG_DEBUG,"Sending Reset to %s\n",peer->username);
+				sip_send_reset_to_peer(peer);
+			} else {
+				if (!strcmp(peer->username,argv[3])) {
+				ast_log(LOG_DEBUG,"Sending Reset to %s\n",peer->username);
+				sip_send_reset_to_peer(peer);
+				}
+			}
+}
+	ast_mutex_unlock(&peerl.lock);
+	return RESULT_SUCCESS;
+}
+
+
 static char *regstate2str(int regstate)
 {
 	switch(regstate) {
@@ -4317,6 +4402,12 @@
 "Usage: sip no debug\n"
 "       Disables dumping of SIP packets for debugging purposes\n";
 
+static char show_reset_phone_peer[] = 
+"Usage: sip reset <extension> or <all>\n"
+"       Resets the SIP phone at given extension\n"
+"       If \'all\' was specified, then all SIP phones will be rebooted\n\n"
+"**Be sure you have updated your syncinfo.xml file!*.\n";
+
 static struct ast_cli_entry  cli_show_users = 
 	{ { "sip", "show", "users", NULL }, sip_show_users, "Show defined SIP users", show_users_usage };
 static struct ast_cli_entry  cli_show_channels =
@@ -4333,7 +4424,8 @@
 	{ { "sip", "debug", NULL }, sip_do_debug, "Enable SIP debugging", debug_usage };
 static struct ast_cli_entry  cli_no_debug =
 	{ { "sip", "no", "debug", NULL }, sip_no_debug, "Disable SIP debugging", no_debug_usage };
-
+static struct ast_cli_entry cli_reset_phone =
+	{ { "sip", "reset", "phone", NULL }, sip_reset_phone_peer, "Sending Reset Info", show_reset_phone_peer };
 
 static int sip_poke_peer_s(void *data)
 {
@@ -6003,6 +6095,7 @@
 		ast_cli_register(&cli_debug);
 		ast_cli_register(&cli_no_debug);
 		ast_cli_register(&cli_inuse_show);
+		ast_cli_register(&cli_reset_phone);
 		sip_rtp.type = type;
 		ast_rtp_proto_register(&sip_rtp);
 		ast_mutex_lock(&peerl.lock);