Sun Feb 16 07:00:01 CET 2003

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -1,3 +1,9 @@
+ -- Make HOLD on SIP make use of asterisk MOH
+ -- Add supervised transfer (tested with Pingtel only)
+ -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
+ -- Preliminary codec 13 support (RFC3389)
+ -- Add app_authenticate for general purpose authentication
+ -- Optimize RTP and smoother
  -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
  -- Fix uninitialized frame pointer in channel.c
  -- Add global variables support under [globals] of extensions.conf

Makefile

@@ -144,7 +144,7 @@ datafiles: all
 			exit 1; \
 		fi; \
 	done
-	for x in sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-*; do \
+	for x in sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-*; do \
 		if grep -q "^%`basename $$x`%" sounds.txt; then \
 			install $$x $(ASTVARLIBDIR)/sounds ; \
 		else \

apps/Makefile

@@ -17,7 +17,8 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.
      app_agi.so app_qcall.so app_adsiprog.so app_getcpeid.so app_milliwatt.so \
      app_zapateller.so app_datetime.so app_setcallerid.so app_festival.so \
      app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \
-     app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so
+     app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \
+     app_authenticate.so
 
 #APPS+=app_sql_postgres.so
 #APPS+=app_sql_odbc.so

apps/app_authenticate.c

@@ -0,0 +1,154 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Execute arbitrary authenticate commands
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/lock.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/app.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <stdlib.h>
+
+#include <pthread.h>
+
+
+static char *tdesc = "Authentication Application";
+
+static char *app = "Authenticate";
+
+static char *synopsis = "Authenticate a user";
+
+static char *descrip =
+"  Authenticate(password[|options]): Requires a user to enter a"
+"given password in order to continue execution.  If the\n"
+"password begins with the '/' character, it is interpreted as\n"
+"a file which contains a list of valid passwords (1 per line).\n"
+"an optional set of opions may be provided by concatenating any\n"
+"of the following letters:\n"
+"     a - Set account code to the password that is entered\n"
+"\n"
+"Returns 0 if the user enters a valid password within three\n"
+"tries, or -1 otherwise (or on hangup).\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int auth_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	int retries;
+	struct localuser *u;
+	char password[256]="";
+	char passwd[256];
+	char *opts;
+	char *prompt;
+	if (!data || !strlen(data)) {
+		ast_log(LOG_WARNING, "Authenticate requires an argument(password)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	if (chan->_state != AST_STATE_UP) {
+		res = ast_answer(chan);
+		if (res) {
+			LOCAL_USER_REMOVE(u);
+			return -1;
+		}
+	}
+	strncpy(password, data, sizeof(password) - 1);
+	opts=strchr(password, '|');
+	if (opts) {
+		*opts = 0;
+		opts++;
+	} else
+		opts = "";
+	/* Start asking for password */
+	prompt = "agent-pass";
+	for (retries = 0; retries < 3; retries++) {
+		res = ast_app_getdata(chan, prompt, passwd, sizeof(passwd) - 2, 0);
+		if (res < 0)
+			break;
+		res = 0;
+		if (password[0] == '/') {
+			/* Compare against a file */
+			char tmp[80];
+			FILE *f;
+			f = fopen(password, "r");
+			if (f) {
+				char buf[256] = "";
+				while(!feof(f)) {
+					fgets(buf, sizeof(buf), f);
+					if (!feof(f) && strlen(buf)) {
+						buf[strlen(buf) - 1] = '\0';
+						if (strlen(buf) && !strcmp(passwd, buf))
+							break;
+					}
+				}
+				fclose(f);
+				if (strlen(buf) && !strcmp(passwd, buf))
+					break;
+			} else 
+				ast_log(LOG_WARNING, "Unable to open file '%s' for authentication: %s\n", password, strerror(errno));
+		} else {
+			/* Compare against a fixed password */
+			if (!strcmp(passwd, password)) 
+				break;
+		}
+		prompt="auth-incorrect";
+	}
+	if ((retries < 3) && !res) {
+		if (strchr(opts, 'a')) 
+			ast_cdr_setaccount(chan, passwd);
+	} else {
+		if (!res)
+			res = ast_streamfile(chan, "vm-goodbye", chan->language);
+		if (!res)
+			res = ast_waitstream(chan, "");
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, auth_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

channels/chan_mgcp.c

@@ -131,6 +131,7 @@ struct mgcp_endpoint {
 	int outgoing;
 	struct ast_channel *owner;
 	struct ast_rtp *rtp;
+	struct sockaddr_in tmpdest;
 	struct mgcp_endpoint *next;
 	struct mgcp_gateway *parent;
 };
@@ -317,6 +318,7 @@ static int mgcp_hangup(struct ast_channel *ast)
 	p->owner = NULL;
 	if (strlen(p->cxident))
 		transmit_connection_del(p);
+	strcpy(p->cxident, "");
 	if (!p->alreadygone && (!p->outgoing || (ast->_state == AST_STATE_UP)))
 		transmit_notify_request(p, "ro", 1);
 	else
@@ -324,8 +326,9 @@ static int mgcp_hangup(struct ast_channel *ast)
 	ast->pvt->pvt = NULL;
 	p->alreadygone = 0;
 	p->outgoing = 0;
-	strcpy(p->cxident, "");
 	strcpy(p->callid, "");
+	/* Reset temporary destination */
+	memset(&p->tmpdest, 0, sizeof(p->tmpdest));
 	if (p->rtp) {
 		ast_rtp_destroy(p->rtp);
 		p->rtp = NULL;
@@ -515,6 +518,7 @@ static struct ast_channel *mgcp_new(struct mgcp_endpoint *i, int state)
 		tmp->pvt->indicate = mgcp_indicate;
 		tmp->pvt->fixup = mgcp_fixup;
 		tmp->pvt->send_digit = mgcp_senddigit;
+		tmp->pvt->bridge = ast_rtp_bridge;
 		if (strlen(i->language))
 			strncpy(tmp->language, i->language, sizeof(tmp->language)-1);
 		i->owner = tmp;
@@ -951,8 +955,15 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_endpoint *p, struct as
 	if (rtp) {
 		ast_rtp_get_peer(rtp, &dest);
 	} else {
-		dest.sin_addr = p->parent->ourip;
-		dest.sin_port = sin.sin_port;
+		if (p->tmpdest.sin_addr.s_addr) {
+			dest.sin_addr = p->tmpdest.sin_addr;
+			dest.sin_port = p->tmpdest.sin_port;
+			/* Reset temporary destination */
+			memset(&p->tmpdest, 0, sizeof(p->tmpdest));
+		} else {
+			dest.sin_addr = p->parent->ourip;
+			dest.sin_port = sin.sin_port;
+		}
 	}
 	printf("We're at %s port %d\n", inet_ntoa(p->parent->ourip), ntohs(sin.sin_port));	
 	snprintf(v, sizeof(v), "v=0\r\n");
@@ -991,6 +1002,12 @@ static int transmit_modify_with_sdp(struct mgcp_endpoint *p, struct ast_rtp *rtp
 	char local[256];
 	char tmp[80];
 	int x;
+	if (!strlen(p->cxident) && rtp) {
+		/* We don't have a CXident yet, store the destination and
+		   wait a bit */
+		ast_rtp_get_peer(rtp, &p->tmpdest);
+		return 0;
+	}
 	snprintf(local, sizeof(local), "p:20");
 	for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
 		if (p->capability & x) {
@@ -1003,6 +1020,7 @@ static int transmit_modify_with_sdp(struct mgcp_endpoint *p, struct ast_rtp *rtp
 	add_header(&resp, "L", local);
 	add_header(&resp, "M", "sendrecv");
 	add_header(&resp, "X", p->txident);
+	add_header(&resp, "I", p->cxident);
 	add_header(&resp, "S", "");
 	add_sdp(&resp, p, rtp);
 	p->lastout = oseq;
@@ -1278,8 +1296,14 @@ static int mgcpsock_read(int *id, int fd, short events, void *ignore)
 		p = find_endpoint(NULL, ident, &sin);
 		if (p) {
 			handle_response(p, result, ident);
-			if ((c = get_header(&req, "I")))
-				strncpy(p->cxident, c, sizeof(p->cxident) - 1);
+			if ((c = get_header(&req, "I"))) {
+				if (strlen(c)) {
+					strncpy(p->cxident, c, sizeof(p->cxident) - 1);
+					if (p->tmpdest.sin_addr.s_addr) {
+						transmit_modify_with_sdp(p, NULL);
+					}
+				}
+			}
 			if (req.lines)
 				process_sdp(p, &req);
 		}
@@ -1483,6 +1507,31 @@ struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
 	return gw;
 }
 
+static struct ast_rtp *mgcp_get_rtp_peer(struct ast_channel *chan)
+{
+	struct mgcp_endpoint *p;
+	p = chan->pvt->pvt;
+	if (p && p->rtp)
+		return p->rtp;
+	return NULL;
+}
+
+static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp)
+{
+	struct mgcp_endpoint *p;
+	p = chan->pvt->pvt;
+	if (p) {
+		transmit_modify_with_sdp(p, rtp);
+		return 0;
+	}
+	return -1;
+}
+
+static struct ast_rtp_protocol mgcp_rtp = {
+	get_rtp_info: mgcp_get_rtp_peer,
+	set_rtp_peer: mgcp_set_rtp_peer,
+};
+
 int load_module()
 {
 	struct ast_config *cfg;
@@ -1590,6 +1639,8 @@ int load_module()
 		ast_destroy(cfg);
 		return -1;
 	}
+	mgcp_rtp.type = type;
+	ast_rtp_proto_register(&mgcp_rtp);
 	ast_cli_register(&cli_show_endpoints);
 	/* And start the monitor for the first time */
 	restart_monitor();

channels/chan_sip.c

@@ -32,6 +32,7 @@
 #include <asterisk/cli.h>
 #include <asterisk/md5.h>
 #include <asterisk/app.h>
+#include <asterisk/musiconhold.h>
 #include <sys/socket.h>
 #include <sys/ioctl.h>
 #include <net/if.h>
@@ -47,8 +48,12 @@
 /* #define VOCAL_DATA_HACK */
 
 #define SIPDUMPER
-#define DEFAULT_EXPIREY 120
-#define MAX_EXPIREY     3600
+#define DEFAULT_DEFAULT_EXPIREY 120
+#define DEFAULT_MAX_EXPIREY     3600
+
+static int max_expirey = DEFAULT_MAX_EXPIREY;
+static int default_expirey = DEFAULT_DEFAULT_EXPIREY;
+
 #define DEFAULT_MAXMS		2000		/* Must be faster than 2 seconds by default */
 
 #define DEFAULT_MAXMS		2000		/* Must be faster than 2 seconds by default */
@@ -142,6 +147,7 @@ static struct sip_pvt {
 	char refer_to[AST_MAX_EXTENSION];	/* Place to store REFER-TO extension */
 	char referred_by[AST_MAX_EXTENSION];/* Place to store REFERRED-BY extension */
 	char refer_contact[AST_MAX_EXTENSION];/* Place to store Contact info from a REFER extension */
+	struct sip_pvt *refer_call;			/* Call we are referring */
 	char record_route[256];
 	char record_route_info[256];
 	char remote_party_id[256];
@@ -728,6 +734,7 @@ static int sip_indicate(struct ast_channel *ast, int condition)
 }
 
 
+#if 0
 static int sip_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
 {
 	struct sip_pvt *p0, *p1;
@@ -742,6 +749,7 @@ static int sip_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags,
 	ast_pthread_mutex_lock(&c1->lock);
 	p0 = c0->pvt->pvt;
 	p1 = c1->pvt->pvt;
+	ast_log(LOG_DEBUG, "Reinvite? %s: %s, %s: %s\n", c0->name, p0->canreinvite ? "yes" : "no", c1->name, p1->canreinvite ? "yes" : "no");
 	if (!p0->canreinvite || !p1->canreinvite) {
 		/* Not gonna support reinvite */
 		ast_pthread_mutex_unlock(&c0->lock);
@@ -796,6 +804,7 @@ static int sip_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags,
 	}
 	return -1;
 }
+#endif
 
 static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 {
@@ -808,7 +817,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 			tmp->nativeformats = capability;
 		fmt = ast_best_codec(tmp->nativeformats);
 		if (title)
-			snprintf(tmp->name, sizeof(tmp->name), "SIP/%s", title);
+			snprintf(tmp->name, sizeof(tmp->name), "SIP/%s-%04x", title, rand() & 0xffff);
 		else
 			snprintf(tmp->name, sizeof(tmp->name), "SIP/%s:%d", inet_ntoa(i->sa.sin_addr), ntohs(i->sa.sin_port));
 		tmp->type = type;
@@ -830,7 +839,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 		tmp->pvt->indicate = sip_indicate;
 		tmp->pvt->fixup = sip_fixup;
 		tmp->pvt->send_digit = sip_senddigit;
-		tmp->pvt->bridge = sip_bridge;
+		tmp->pvt->bridge = ast_rtp_bridge;
 		if (strlen(i->language))
 			strncpy(tmp->language, i->language, sizeof(tmp->language)-1);
 		i->owner = tmp;
@@ -1087,7 +1096,7 @@ static int sip_register(char *value, int lineno)
 		if (secret)
 			strncpy(reg->secret, secret, sizeof(reg->secret)-1);
 		reg->expire = -1;
-		reg->refresh = DEFAULT_EXPIREY;
+		reg->refresh = default_expirey;
 		reg->addr.sin_family = AF_INET;
 		memcpy(&reg->addr.sin_addr, hp->h_addr, sizeof(&reg->addr.sin_addr));
 		reg->addr.sin_port = porta ? htons(atoi(porta)) : htons(DEFAULT_SIP_PORT);
@@ -1237,11 +1246,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		ast_log(LOG_WARNING, "No compatible codecs!\n");
 		return -1;
 	}
-	if (p->owner && !(p->owner->nativeformats & p->capability)) {
-		ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %d and not %d\n", p->capability, p->owner->nativeformats);
-		p->owner->nativeformats = p->capability;
-		ast_set_read_format(p->owner, p->owner->readformat);
-		ast_set_write_format(p->owner, p->owner->writeformat);
+	if (p->owner) {
+		if (p->owner->nativeformats & p->capability) {
+			ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %d and not %d\n", p->capability, p->owner->nativeformats);
+			p->owner->nativeformats = p->capability;
+			ast_set_read_format(p->owner, p->owner->readformat);
+			ast_set_write_format(p->owner, p->owner->writeformat);
+		}
+		if (p->owner->bridge) {
+			/* Turn on/off music on hold if we are holding/unholding */
+			if (sin.sin_addr.s_addr) {
+				ast_moh_stop(p->owner->bridge);
+			} else {
+				ast_moh_start(p->owner->bridge, NULL);
+			}
+		}
 	}
 	return 0;
 	
@@ -1830,7 +1849,7 @@ static int transmit_register(struct sip_registry *r, char *cmd, char *auth)
 	if (auth) 
 		add_header(&req, "Authorization", auth);
 
-	snprintf(tmp, sizeof(tmp), "%d", DEFAULT_EXPIREY);
+	snprintf(tmp, sizeof(tmp), "%d", default_expirey);
 	add_header(&req, "Expires", tmp);
 	add_header(&req, "Event", "registration");
 	copy_request(&p->initreq, &req);
@@ -1933,8 +1952,8 @@ static int parse_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_req
 		strcpy(p->username, "");
 	if (p->expire > -1)
 		ast_sched_del(sched, p->expire);
-	if ((expirey < 1) || (expirey > MAX_EXPIREY))
-		expirey = DEFAULT_EXPIREY;
+	if ((expirey < 1) || (expirey > max_expirey))
+		expirey = max_expirey;
 	p->expire = ast_sched_add(sched, expirey * 1000, expire_register, p);
 	pvt->expirey = expirey;
 	if (memcmp(&p->addr, &oldsin, sizeof(oldsin))) {
@@ -2130,7 +2149,9 @@ static int get_refer_info(struct sip_pvt *p, struct sip_request *oreq)
 	char tmp2[256] = "", *c2, *a2;
 	char tmp3[256];
 	char tmp4[256];
+	char tmp5[256] = "";		/* CallID to replace */
 	struct sip_request *req;
+	struct sip_pvt *p2;
 	
 	req = oreq;
 	if (!req)
@@ -2151,32 +2172,81 @@ static int get_refer_info(struct sip_pvt *p, struct sip_request *oreq)
 	}
 	c += 4;
 	c2 += 4;
-	if ((a = strchr(c, '@')) || (a = strchr(c, ';'))) {
+	if ((a = strchr(c, '?'))) {
+		/* Search for arguemnts */
 		*a = '\0';
+		a++;
+		if (!strncasecmp(a, "REPLACES=", strlen("REPLACES="))) {
+			strncpy(tmp5, a + strlen("REPLACES="), sizeof(tmp5) - 1);
+			if ((a = strchr(tmp5, '%'))) {
+				/* Yuck!  Pingtel converts the '@' to a %40, icky icky!  Convert
+				   back to an '@' */
+				if ((a[1] == '4') && (a[2] == '0')) {
+					*a = '@';
+					memmove(a + 1, a+3, strlen(a + 3));
+				}
+			}
+			if ((a = strchr(tmp5, '%'))) 
+				*a = '\0';
+		}
 	}
-	if ((a2 = strchr(c2, '@')) || (a2 = strchr(c2, ';'))) {	
+	
+	if ((a = strchr(c, '@')))
+		*a = '\0';
+	if ((a = strchr(c, ';'))) 
+		*a = '\0';
+	
+
+	if ((a2 = strchr(c2, '@')))
 		*a2 = '\0';
-	}
+
+	if ((a2 = strchr(c2, ';'))) 
+		*a2 = '\0';
+	
 	
 	if (sipdebug)
 		ast_verbose("Looking for %s in %s\n", c, p->context);
 		ast_verbose("Looking for %s in %s\n", c2, p->context);
-	
-	if (ast_exists_extension(NULL, p->context, c, 1, NULL) && ast_exists_extension(NULL, p->context, c2, 1, NULL)) {
-		if (!oreq)
-			ast_log(LOG_DEBUG,"Something is wrong with this line.\n");	//This line is ignored for some reason....
-			ast_log(LOG_DEBUG,"Assigning Extension %s to REFER-TO\n", c);
-			ast_log(LOG_DEBUG,"Assigning Extension %s to REFERRED-BY\n", c2);
-			ast_log(LOG_DEBUG,"Assigning Contact Info %s to REFER_CONTACT\n", tmp3);
-			ast_log(LOG_DEBUG,"Assigning Remote-Party-ID Info %s to REMOTE_PARTY_ID\n",tmp4);
-			strncpy(p->refer_to, c, sizeof(p->refer_to) - 1);
-			strncpy(p->referred_by, c2, sizeof(p->referred_by) - 1);
-			strncpy(p->refer_contact, tmp3, sizeof(p->refer_contact) - 1);
-			strncpy(p->remote_party_id, tmp4, sizeof(p->remote_party_id) - 1);
+		
+	if (strlen(tmp5)) {	
+		/* This is a supervised transfer */
+		ast_log(LOG_DEBUG,"Assigning Replace-Call-ID Info %s to REPLACE_CALL_ID\n",tmp5);
+		
+		strncpy(p->refer_to, "", sizeof(p->refer_to) - 1);
+		strncpy(p->referred_by, "", sizeof(p->referred_by) - 1);
+		strncpy(p->refer_contact, "", sizeof(p->refer_contact) - 1);
+		strncpy(p->remote_party_id, "", sizeof(p->remote_party_id) - 1);
+		p->refer_call = NULL;
+		ast_pthread_mutex_lock(&iflock);
+		/* Search interfaces and find the match */
+		p2 = iflist;
+		while(p2) {
+			if (!strcmp(p2->callid, tmp5)) {
+				/* Go ahead and lock it before returning */
+				ast_pthread_mutex_lock(&p2->lock);
+				p->refer_call = p2;
+				break;
+			}
+			p2 = p2->next;
+		}
+		ast_pthread_mutex_unlock(&iflock);
+		if (p->refer_call)
 			return 0;
-	}
-
-	if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
+		else
+			ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'\n", tmp5);
+	} else if (ast_exists_extension(NULL, p->context, c, 1, NULL) && ast_exists_extension(NULL, p->context, c2, 1, NULL)) {
+		/* This is an unsupervised transfer */
+		ast_log(LOG_DEBUG,"Assigning Extension %s to REFER-TO\n", c);
+		ast_log(LOG_DEBUG,"Assigning Extension %s to REFERRED-BY\n", c2);
+		ast_log(LOG_DEBUG,"Assigning Contact Info %s to REFER_CONTACT\n", tmp3);
+		ast_log(LOG_DEBUG,"Assigning Remote-Party-ID Info %s to REMOTE_PARTY_ID\n",tmp4);
+		strncpy(p->refer_to, c, sizeof(p->refer_to) - 1);
+		strncpy(p->referred_by, c2, sizeof(p->referred_by) - 1);
+		strncpy(p->refer_contact, tmp3, sizeof(p->refer_contact) - 1);
+		strncpy(p->remote_party_id, tmp4, sizeof(p->remote_party_id) - 1);
+		p->refer_call = NULL;
+		return 0;
+	} else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
 		return 1;
 	}
 
@@ -2735,7 +2805,7 @@ retrylock:
 				if (r->expire != -1)
 					ast_sched_del(sched, r->expire);
 				expires=atoi(get_header(req, "expires"));
-				if (!expires) expires=DEFAULT_EXPIREY;
+				if (!expires) expires=default_expirey;
 					r->expire=ast_sched_add(sched, (expires-2)*1000, sip_reregister, r); 
 
 			}
@@ -2879,6 +2949,37 @@ static int determine_firstline_parts( struct sip_request *req ) {
   return 1;
 }
 
+static int attempt_transfer(struct sip_pvt *p1, struct sip_pvt *p2)
+{
+	if (!p1->owner || !p2->owner) {
+		ast_log(LOG_WARNING, "Transfer attempted without dual ownership?\n");
+		return -1;
+	}
+	if (p1->owner->bridge) {
+		if (p2->owner->bridge)
+			ast_moh_stop(p2->owner->bridge);
+		ast_moh_stop(p1->owner->bridge);
+		ast_moh_stop(p1->owner);
+		ast_moh_stop(p2->owner);
+		if (ast_channel_masquerade(p2->owner, p1->owner->bridge)) {
+			ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", p2->owner->name, p1->owner->bridge->name);
+			return -1;
+		}
+	} else if (p2->owner->bridge) {
+		ast_moh_stop(p2->owner->bridge);
+		ast_moh_stop(p2->owner);
+		ast_moh_stop(p1->owner);
+		if (ast_channel_masquerade(p1->owner, p2->owner->bridge)) {
+			ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", p1->owner->name, p2->owner->bridge->name);
+			return -1;
+		}
+	} else {
+		ast_log(LOG_NOTICE, "Transfer attempted with no bridged calls to transfer\n");
+		return -1;
+	}
+	return 0;
+}
+
 static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin)
 {
 	struct sip_request resp;
@@ -3048,16 +3149,23 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 			transmit_response_with_allow(p, "404 Not Found", req);
 		else if (res > 0)
 			transmit_response_with_allow(p, "484 Address Incomplete", req);
-		else
+		else {
 			transmit_response(p, "202 Accepted", req);
-		ast_log(LOG_DEBUG,"202 Accepted\n");
-		c = p->owner;
-		if (c) {
-			transfer_to = c->bridge;
-			if (transfer_to)
-				ast_async_goto(transfer_to,"", p->refer_to,1, 1);
+			if (p->refer_call) {
+				ast_log(LOG_DEBUG,"202 Accepted (supervised)\n");
+				attempt_transfer(p, p->refer_call);
+				ast_pthread_mutex_unlock(&p->refer_call->lock);
+				p->refer_call = NULL;
+			} else {
+				ast_log(LOG_DEBUG,"202 Accepted (blind)\n");
+				c = p->owner;
+				if (c) {
+					transfer_to = c->bridge;
+					if (transfer_to)
+						ast_async_goto(transfer_to,"", p->refer_to,1, 1);
+				}
+			}
 		}
-			
 	} else if (!strcasecmp(cmd, "CANCEL") || !strcasecmp(cmd, "BYE")) {
 		copy_request(&p->initreq, req);
 		p->alreadygone = 1;
@@ -3140,7 +3248,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
 		/* Must have at least two headers */
 		return 1;
 	}
-	/* Process request, with iflock held */
+	/* Process request, with netlock held */
 	ast_pthread_mutex_lock(&netlock);
 	p = find_call(&req, &sin);
 	if (p) {
@@ -3495,6 +3603,8 @@ static struct sip_peer *build_peer(char *name, struct ast_variable *v)
 			peer->expirey = expirey;
 		}
 		peer->capability = capability;
+		/* Assume can reinvite */
+		peer->canreinvite = 1;
 		while(v) {
 			if (!strcasecmp(v->name, "secret")) 
 				strncpy(peer->secret, v->value, sizeof(peer->secret)-1);
@@ -3619,6 +3729,14 @@ static int reload_config()
 			strncpy(context, v->value, sizeof(context)-1);
 		} else if (!strcasecmp(v->name, "language")) {
 			strncpy(language, v->value, sizeof(language)-1);
+		} else if (!strcasecmp(v->name, "maxexpirey")) {
+			max_expirey = atoi(v->value);
+			if (max_expirey < 1)
+				max_expirey = DEFAULT_MAX_EXPIREY;
+		} else if (!strcasecmp(v->name, "defaultexpirey")) {
+			default_expirey = atoi(v->value);
+			if (default_expirey < 1)
+				default_expirey = DEFAULT_DEFAULT_EXPIREY;
 		} else if (!strcasecmp(v->name, "bindaddr")) {
 			if (!(hp = gethostbyname(v->value))) {
 				ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
@@ -3743,6 +3861,31 @@ static int reload_config()
 	return 0;
 }
 
+static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
+{
+	struct sip_pvt *p;
+	p = chan->pvt->pvt;
+	if (p && p->rtp && p->canreinvite)
+		return p->rtp;
+	return NULL;
+}
+
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp)
+{
+	struct sip_pvt *p;
+	p = chan->pvt->pvt;
+	if (p) {
+		transmit_reinvite_with_sdp(p, rtp);
+		return 0;
+	}
+	return -1;
+}
+
+static struct ast_rtp_protocol sip_rtp = {
+	get_rtp_info: sip_get_rtp_peer,
+	set_rtp_peer: sip_set_rtp_peer,
+};
+
 int load_module()
 {
 	int res;
@@ -3761,6 +3904,8 @@ int load_module()
 		ast_cli_register(&cli_show_registry);
 		ast_cli_register(&cli_debug);
 		ast_cli_register(&cli_no_debug);
+		sip_rtp.type = type;
+		ast_rtp_proto_register(&sip_rtp);
 		sched = sched_context_create();
 		if (!sched) {
 			ast_log(LOG_WARNING, "Unable to create schedule context\n");

configs/sip.conf.sample

@@ -7,6 +7,8 @@ bindaddr = 0.0.0.0		; Address to bind to
 context = default		; Default for incoming calls
 ;tos=lowdelay
 ;tos=184
+;maxexpirey=3600		; Max length of incoming registration we allow
+;defaultexpirey=120		; Default length of incoming/outoing registration
 
 ;[snomsip]
 ;type=friend

frame.c

@@ -35,10 +35,12 @@ struct ast_smoother {
 	int size;
 	int format;
 	int readdata;
+	int optimizablestream;
 	float samplesperbyte;
 	struct ast_frame f;
 	char data[SMOOTHER_SIZE];
 	char framedata[SMOOTHER_SIZE + AST_FRIENDLY_OFFSET];
+	struct ast_frame *opt;
 	int len;
 };
 
@@ -76,6 +78,28 @@ int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
 		ast_log(LOG_WARNING, "Out of smoother space\n");
 		return -1;
 	}
+	if ((f->datalen == s->size) && !s->opt) {
+		if (!s->len) {
+			/* Optimize by sending the frame we just got
+			   on the next read, thus eliminating the douple
+			   copy */
+			s->opt = f;
+			return 0;
+		} else {
+			s->optimizablestream++;
+			if (s->optimizablestream > 10) {
+				/* For the past 10 rounds, we have input and output
+				   frames of the correct size for this smoother, yet
+				   we were unable to optimize because there was still
+				   some cruft left over.  Lets just drop the cruft so
+				   we can move to a fully optimized path */
+				s->len = 0;
+				s->opt = f;
+				return 0;
+			}
+		}
+	} else 
+		s->optimizablestream = 0;
 	memcpy(s->data + s->len, f->data, f->datalen);
 	s->len += f->datalen;
 	return 0;
@@ -83,6 +107,15 @@ int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
 
 struct ast_frame *ast_smoother_read(struct ast_smoother *s)
 {
+	struct ast_frame *opt;
+
+	/* IF we have an optimization frame, send it */
+	if (s->opt) {
+		opt = s->opt;
+		s->opt = NULL;
+		return opt;
+	}
+
 	/* Make sure we have enough data */
 	if (s->len < s->size) {
 		return NULL;

include/asterisk/rtp.h

@@ -17,6 +17,7 @@
 #include <asterisk/frame.h>
 #include <asterisk/io.h>
 #include <asterisk/sched.h>
+#include <asterisk/channel.h>
 
 #include <netinet/in.h>
 
@@ -24,6 +25,14 @@
 extern "C" {
 #endif
 
+struct ast_rtp_protocol {
+	struct ast_rtp *(*get_rtp_info)(struct ast_channel *chan);				/* Get RTP struct, or NULL if unwilling to transfer */
+	int (*set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer);	/* Set RTP peer */
+	int (*get_rtp_willing)(struct ast_channel *chan);		/* Willing to native bridge */
+	char *type;
+	struct ast_rtp_protocol *next;
+};
+
 struct ast_rtp;
 
 typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
@@ -58,6 +67,12 @@ int rtp2ast(int id);
 
 char *ast2rtpn(int id);
 
+int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc);
+
+int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
+
+void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif

rtp.c

@@ -30,6 +30,7 @@
 #include <asterisk/logger.h>
 #include <asterisk/options.h>
 #include <asterisk/channel.h>
+#include <asterisk/channel_pvt.h>
 
 #define TYPE_SILENCE	 0x2
 #define TYPE_HIGH	 0x0
@@ -47,6 +48,7 @@ struct ast_rtp {
 	unsigned int lastts;
 	unsigned int lastrxts;
 	int lasttxformat;
+	int lastrxformat;
 	int dtmfcount;
 	struct sockaddr_in us;
 	struct sockaddr_in them;
@@ -61,6 +63,8 @@ struct ast_rtp {
 	ast_rtp_callback callback;
 };
 
+static struct ast_rtp_protocol *protos = NULL;
+
 int ast_rtp_fd(struct ast_rtp *rtp)
 {
 	return rtp->s;
@@ -151,6 +155,49 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
 	return f;
 }
 
+static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
+{
+	struct ast_frame *f = NULL;
+	/* Convert comfort noise into audio with various codecs.  Unfortunately this doesn't
+	   totally help us out becuase we don't have an engine to keep it going and we are not
+	   guaranteed to have it every 20ms or anything */
+#if 0
+	printf("RFC3389: %d bytes, format is %d\n", len, rtp->lastrxformat);
+#endif	
+	ast_log(LOG_NOTICE, "RFC3389 support incomplete.  Turn off on client if possible\n");
+	if (!rtp->lastrxformat)
+		return 	NULL;
+	switch(rtp->lastrxformat) {
+	case AST_FORMAT_ULAW:
+		rtp->f.frametype = AST_FRAME_VOICE;
+		rtp->f.subclass = AST_FORMAT_ULAW;
+		rtp->f.datalen = 160;
+		rtp->f.samples = 160;
+		memset(rtp->f.data, 0x7f, rtp->f.datalen);
+		f = &rtp->f;
+		break;
+	case AST_FORMAT_ALAW:
+		rtp->f.frametype = AST_FRAME_VOICE;
+		rtp->f.subclass = AST_FORMAT_ALAW;
+		rtp->f.datalen = 160;
+		rtp->f.samples = 160;
+		memset(rtp->f.data, 0x7e, rtp->f.datalen); /* XXX Is this right? XXX */
+		f = &rtp->f;
+		break;
+	case AST_FORMAT_SLINEAR:
+		rtp->f.frametype = AST_FRAME_VOICE;
+		rtp->f.subclass = AST_FORMAT_SLINEAR;
+		rtp->f.datalen = 320;
+		rtp->f.samples = 160;
+		memset(rtp->f.data, 0x00, rtp->f.datalen);
+		f = &rtp->f;
+		break;
+	default:
+		ast_log(LOG_NOTICE, "Don't know how to handle RFC3389 for receive codec %d\n", rtp->lastrxformat);
+	}
+	return f;
+}
+
 static struct ast_frame *process_type121(struct ast_rtp *rtp, unsigned char *data, int len)
 {
 	char resp = 0;
@@ -247,6 +294,8 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 		} else if (payloadtype == 100) {
 			/* CISCO's notso proprietary DTMF bridge */
 			f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
+		} else if (payloadtype == 13) {
+			f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
 		} else {
 			ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
 		}
@@ -254,7 +303,8 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 			return f;
 		else
 			return &null_frame;
-	}
+	} else
+		rtp->lastrxformat = rtp->f.subclass;
 
 	if (!rtp->lastrxts)
 		rtp->lastrxts = timestamp;
@@ -651,3 +701,155 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
 		
 	return 0;
 }
+
+void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
+{
+	struct ast_rtp_protocol *cur, *prev;
+	cur = protos;
+	prev = NULL;
+	while(cur) {
+		if (cur == proto) {
+			if (prev)
+				prev->next = proto->next;
+			else
+				protos = proto->next;
+			return;
+		}
+		prev = cur;
+		cur = cur->next;
+	}
+}
+
+int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
+{
+	struct ast_rtp_protocol *cur;
+	cur = protos;
+	while(cur) {
+		if (cur->type == proto->type) {
+			ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
+			return -1;
+		}
+		cur = cur->next;
+	}
+	proto->next = protos;
+	protos = proto;
+	return 0;
+}
+
+static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
+{
+	struct ast_rtp_protocol *cur;
+	cur = protos;
+	while(cur) {
+		if (cur->type == chan->type) {
+			return cur;
+		}
+		cur = cur->next;
+	}
+	return NULL;
+}
+
+int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
+{
+	struct ast_frame *f;
+	struct ast_channel *who, *cs[3];
+	struct ast_rtp *p0, *p1;
+	struct ast_rtp_protocol *pr0, *pr1;
+	void *pvt0, *pvt1;
+	int to;
+
+	/* XXX Wait a half a second for things to settle up 
+			this really should be fixed XXX */
+	ast_autoservice_start(c0);
+	ast_autoservice_start(c1);
+	usleep(500000);
+	ast_autoservice_stop(c0);
+	ast_autoservice_stop(c1);
+
+	/* if need DTMF, cant native bridge */
+	if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
+		return -2;
+	ast_pthread_mutex_lock(&c0->lock);
+	ast_pthread_mutex_lock(&c1->lock);
+	pr0 = get_proto(c0);
+	pr1 = get_proto(c1);
+	if (!pr0) {
+		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
+		ast_pthread_mutex_unlock(&c0->lock);
+		ast_pthread_mutex_unlock(&c1->lock);
+		return -1;
+	}
+	if (!pr1) {
+		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
+		ast_pthread_mutex_unlock(&c0->lock);
+		ast_pthread_mutex_unlock(&c1->lock);
+		return -1;
+	}
+	pvt0 = c0->pvt->pvt;
+	pvt1 = c1->pvt->pvt;
+	p0 = pr0->get_rtp_info(c0);
+	p1 = pr1->get_rtp_info(c1);
+	if (!p0 || !p1) {
+		/* Somebody doesn't want to play... */
+		ast_pthread_mutex_unlock(&c0->lock);
+		ast_pthread_mutex_unlock(&c1->lock);
+		return -2;
+	}
+	if (pr0->set_rtp_peer(c0, p1)) 
+		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
+	if (pr1->set_rtp_peer(c1, p0)) 
+		ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
+	ast_pthread_mutex_unlock(&c0->lock);
+	ast_pthread_mutex_unlock(&c1->lock);
+	cs[0] = c0;
+	cs[1] = c1;
+	cs[2] = NULL;
+	for (;;) {
+		if ((c0->pvt->pvt != pvt0)  ||
+			(c1->pvt->pvt != pvt1) ||
+			(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
+				ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
+				if (c0->pvt->pvt == pvt0) {
+					if (pr0->set_rtp_peer(c0, NULL)) 
+						ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
+				}
+				if (c1->pvt->pvt == pvt1) {
+					if (pr1->set_rtp_peer(c1, NULL)) 
+						ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
+				}
+				/* Tell it to try again later */
+				return -3;
+		}
+		to = -1;
+		who = ast_waitfor_n(cs, 2, &to);
+		if (!who) {
+			ast_log(LOG_DEBUG, "Ooh, empty read...\n");
+			continue;
+		}
+		f = ast_read(who);
+		if (!f || ((f->frametype == AST_FRAME_DTMF) &&
+				   (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
+			       ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
+			*fo = f;
+			*rc = who;
+			ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
+			if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) {
+				if (pr0->set_rtp_peer(c0, NULL)) 
+					ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
+			}
+			if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) {
+				if (pr1->set_rtp_peer(c1, NULL)) 
+					ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
+			}
+			/* That's all we needed */
+			return 0;
+		} else 
+			ast_frfree(f);
+		/* Swap priority not that it's a big deal at this point */
+		cs[2] = cs[0];
+		cs[0] = cs[1];
+		cs[1] = cs[2];
+		
+	}
+	return -1;
+}

sounds.txt

@@ -14,6 +14,8 @@
 
 %agent-user.gsm%Agent login.  Please enter your agent number followed by the pound key.
 
+%auth-incorrect.gsm%Login incorrect.  Please enter your password followed by the pound key.
+
 %beep.gsm%(this is a simple beep tone)
 
 %conf-getconfno.gsm%Please enter your conference number followed by the pound key.