dom mar 2 20:52:10 CET 2003

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@629 f38db490-d61c-443f-a65b-d21fe96a405b

CHANGES

@@ -1,3 +1,4 @@
+ -- Add experimental "IAX2" protocol
  -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
  -- Choose best priority from codec from allow/disallow
  -- Reject SIP calls to self

Makefile

@@ -40,7 +40,7 @@ PROC=$(shell uname -m)
 
 DEBUG=-g #-pg
 INCLUDE=-Iinclude -I../include
-CFLAGS=-pipe  -Wall -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
+CFLAGS=-pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
 #CFLAGS+=-O6
 CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
 CFLAGS+=$(shell if uname -m | grep -q ppc; then echo "-fsigned-char"; fi)

channels/Makefile

@@ -14,7 +14,7 @@
 CHANNEL_LIBS=chan_modem.so chan_iax.so chan_sip.so \
 	     chan_modem_aopen.so chan_oss.so \
              chan_modem_bestdata.so chan_modem_i4l.so \
-             chan_agent.so chan_mgcp.so
+             chan_agent.so chan_mgcp.so chan_iax2.so
 
 #
 # If you really want VoFR you can have it :-P

channels/chan_iax.c

@@ -871,7 +871,7 @@ static int do_deliver(void *data)
 	return res;
 }
 
-static int handle_error()
+static int handle_error(void)
 {
 	/* XXX Ideally we should figure out why an error occured and then abort those
 	   rather than continuing to try.  Unfortunately, the published interface does
@@ -4279,7 +4279,7 @@ static void *network_thread(void *ignore)
 	return NULL;
 }
 
-static int start_network_thread()
+static int start_network_thread(void)
 {
 	return pthread_create(&netthreadid, NULL, network_thread, NULL);
 }
@@ -4556,7 +4556,7 @@ void prune_peers(void){
 }
 
 
-int set_config(char *config_file, struct sockaddr_in* sin){
+static int set_config(char *config_file, struct sockaddr_in* sin){
 	struct ast_config *cfg;
 	int capability=iax_capability;
 	struct ast_variable *v;

channels/chan_oss.c

@@ -819,7 +819,7 @@ static int console_sendtext(int fd, int argc, char *argv[])
 	if (strlen(text2send))
 		ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
 	strcpy(text2send, "");
-	while(tmparg <= argc) {
+	while(tmparg < argc) {
 		strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send));
 		strncat(text2send, " ", sizeof(text2send) - strlen(text2send));
 	}

channels/iax2.h

@@ -0,0 +1,126 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Implementation of Inter-Asterisk eXchange
+ * 
+ * Copyright (C) 2003, Digium
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#ifndef _ASTERISK_IAX2_H
+#define _ASTERISK_IAX2_H
+
+/* Max version of IAX protocol we support */
+#define AST_IAX2_PROTO_VERSION 2
+
+#define AST_IAX2_MAX_CALLS 32768
+
+#define AST_FLAG_FULL		0x8000
+
+#define AST_FLAG_RETRANS	0x8000
+
+#define AST_FLAG_SC_LOG		0x80
+
+#define AST_MAX_SHIFT		0x1F
+
+#define AST_IAX2_WINDOW		256
+
+/* Subclass for AST_FRAME_IAX */
+#define AST_IAX2_COMMAND_NEW		1
+#define AST_IAX2_COMMAND_PING	2
+#define AST_IAX2_COMMAND_PONG	3
+#define AST_IAX2_COMMAND_ACK		4
+#define AST_IAX2_COMMAND_HANGUP	5
+#define AST_IAX2_COMMAND_REJECT	6
+#define AST_IAX2_COMMAND_ACCEPT	7
+#define AST_IAX2_COMMAND_AUTHREQ	8
+#define AST_IAX2_COMMAND_AUTHREP	9
+#define AST_IAX2_COMMAND_INVAL	10
+#define AST_IAX2_COMMAND_LAGRQ	11
+#define AST_IAX2_COMMAND_LAGRP	12
+#define AST_IAX2_COMMAND_REGREQ	13	/* Registration request */
+#define AST_IAX2_COMMAND_REGAUTH	14	/* Registration authentication required */
+#define AST_IAX2_COMMAND_REGACK	15	/* Registration accepted */
+#define AST_IAX2_COMMAND_REGREJ	16	/* Registration rejected */
+#define AST_IAX2_COMMAND_REGREL	17	/* Force release of registration */
+#define AST_IAX2_COMMAND_VNAK	18	/* If we receive voice before valid first voice frame, send this */
+#define AST_IAX2_COMMAND_DPREQ	19	/* Request status of a dialplan entry */
+#define AST_IAX2_COMMAND_DPREP	20	/* Request status of a dialplan entry */
+#define AST_IAX2_COMMAND_DIAL	21	/* Request a dial on channel brought up TBD */
+#define AST_IAX2_COMMAND_TXREQ	22	/* Transfer Request */
+#define AST_IAX2_COMMAND_TXCNT	23	/* Transfer Connect */
+#define AST_IAX2_COMMAND_TXACC	24	/* Transfer Accepted */
+#define AST_IAX2_COMMAND_TXREADY	25	/* Transfer ready */
+#define AST_IAX2_COMMAND_TXREL	26	/* Transfer release */
+#define AST_IAX2_COMMAND_TXREJ	27	/* Transfer reject */
+#define AST_IAX2_COMMAND_QUELCH	28	/* Stop audio/video transmission */
+#define AST_IAX2_COMMAND_UNQUELCH 29	/* Resume audio/video transmission */
+#define AST_IAX2_COMMAND_POKE    30  /* Like ping, but does not require an open connection */
+#define AST_IAX2_COMMAND_PAGE	31	/* Paging description */
+#define AST_IAX2_COMMAND_MWI	32	/* Stand-alone message waiting indicator */
+#define AST_IAX2_COMMAND_UNSUPPORT	33	/* Unsupported message received */
+
+#define AST_DEFAULT_REG_EXPIRE  60	/* By default require re-registration once per minute */
+
+#define AST_DEFAULT_IAX_PORTNO	4569
+
+/* IAX Information elements */
+#define IAX_IE_CALLED_NUMBER		1		/* Number/extension being called - string */
+#define IAX_IE_CALLING_NUMBER		2		/* Calling number - string */
+#define IAX_IE_CALLING_ANI			3		/* Calling number ANI for billing  - string */
+#define IAX_IE_CALLING_NAME			4		/* Name of caller - string */
+#define IAX_IE_CALLED_CONTEXT		5		/* Context for number - string */
+#define IAX_IE_USERNAME				6		/* Username (peer or user) for authentication - string */
+#define IAX_IE_PASSWORD				7		/* Password for authentication - string */
+#define IAX_IE_CAPABILITY			8		/* Actual codec capability - unsigned int */
+#define IAX_IE_FORMAT				9		/* Desired codec format - unsigned int */
+#define IAX_IE_LANGUAGE				10		/* Desired language - string */
+#define IAX_IE_VERSION				11		/* Protocol version - short */
+#define IAX_IE_ADSICPE				12		/* CPE ADSI capability - int */
+#define IAX_IE_DNID					13		/* Originally dialed DNID - string */
+#define IAX_IE_AUTHMETHODS			14		/* Authentication method(s) - short */
+#define IAX_IE_CHALLENGE			15		/* Challenge data for MD5/RSA - string */
+#define IAX_IE_MD5_RESULT			16		/* MD5 challenge result - string */
+#define IAX_IE_RSA_RESULT			17		/* RSA challenge result - string */
+#define IAX_IE_APPARENT_ADDR		18		/* Apparent address of peer - struct sockaddr_in */
+#define IAX_IE_REFRESH				19		/* When to refresh registration - short */
+#define IAX_IE_DPSTATUS				20		/* Dialplan status - short */
+#define IAX_IE_CALLNO				21		/* Call number of peer - short */
+#define IAX_IE_CAUSE				22		/* Cause - string */
+
+#define IAX_AUTH_PLAINTEXT			(1 << 0)
+#define IAX_AUTH_MD5				(1 << 1)
+#define IAX_AUTH_RSA				(1 << 2)
+
+#define IAX_DPSTATUS_EXISTS			(1 << 0)
+#define IAX_DPSTATUS_CANEXIST		(1 << 1)
+#define IAX_DPSTATUS_NONEXISTANT	(1 << 2)
+#define IAX_DPSTATUS_IGNOREPAT		(1 << 14)
+#define IAX_DPSTATUS_MATCHMORE		(1 << 15)
+
+/* Full frames are always delivered reliably */
+struct ast_iax2_full_hdr {
+	unsigned short scallno;	/* Source call number -- high bit must be 1 */
+	unsigned short dcallno;	/* Destination call number -- high bit is 1 if retransmission */
+	unsigned int ts;		/* 32-bit timestamp in milliseconds (from 1st transmission) */
+	unsigned short oseqno;	/* Packet number (outgoing) */
+	unsigned short iseqno;	/* Packet number (next incoming expected) */
+	char type;				/* Frame type */
+	unsigned char csub;		/* Compressed subclass */
+	unsigned char iedata[0];
+};
+
+/* Mini header is used only for voice frames -- delivered unreliably */
+struct ast_iax2_mini_hdr {
+	short callno;			/* Source call number -- high bit must be 0 */
+	unsigned short ts;		/* 16-bit Timestamp (high 16 bits from last ast_iax2_full_hdr) */
+							/* Frametype implicitly VOICE_FRAME */
+							/* subclass implicit from last ast_iax2_full_hdr */
+	unsigned char iedata[0];
+};
+
+#endif

codecs/codec_gsm.c

@@ -196,6 +196,8 @@ static struct ast_frame *lintogsm_frameout(struct ast_translator_pvt *tmp)
 
 static void gsm_destroy_stuff(struct ast_translator_pvt *pvt)
 {
+	if (pvt->gsm)
+		gsm_destroy(pvt->gsm);
 	free(pvt);
 	localusecnt--;
 }

pbx.c

@@ -1135,7 +1135,8 @@ int ast_pbx_run(struct ast_channel *c)
 				c->name, c->context, c->exten, c->priority);			
 			if ((res = ast_spawn_extension(c, c->context, c->exten, c->priority, c->callerid))) {
 				/* Something bad happened, or a hangup has been requested. */
-				if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F'))) {
+				if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
+					(res == '*') || (res == '#')) {
 					ast_log(LOG_DEBUG, "Oooh, got something to jump out with ('%c')!\n", res);
 					exten[pos++] = digit = res;
 					break;