o Update to 1.4.19.2;

o move all additional functionality into separate patches and make it
opt in. It has been concern of asterisks devs that the FreeBSD pacakage
adds functionalty not present in the original version, whch could be
confusing.

1 files changed, 225 insertions, 0 deletions

diff --git a/net/asterisk14/files/dtmf_debug.diff b/net/asterisk14/files/dtmf_debug.diff
new file mode 100644
index 000000000000..81205d40dd46
--- /dev/null
+++ b/net/asterisk14/files/dtmf_debug.diff
@@ -0,0 +1,225 @@
+--- channels/chan_sip.c.orig	2008-03-18 16:42:59.000000000 +0200
++++ channels/chan_sip.c	2008-03-18 17:08:34.000000000 +0200
+@@ -3768,6 +3768,7 @@
+ 		ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
+ 	else {
+ 		p->owner = newchan;
++		ast_rtp_set_chan_name(p->rtp, newchan->name);
+ 		ret = 0;
+ 	}
+ 	if (option_debug > 2)
+@@ -4032,6 +4035,7 @@
+ 	if (i->rtp) {
+ 		tmp->fds[0] = ast_rtp_fd(i->rtp);
+ 		tmp->fds[1] = ast_rtcp_fd(i->rtp);
++		ast_rtp_set_chan_id(i->rtp, i->callid);
+ 	}
+ 	if (needvideo && i->vrtp) {
+ 		tmp->fds[2] = ast_rtp_fd(i->vrtp);
+@@ -4059,6 +4063,8 @@
+ 	if (!ast_strlen_zero(i->language))
+ 		ast_string_field_set(tmp, language, i->language);
+ 	i->owner = tmp;
++	ast_rtp_set_chan_name(i->rtp, tmp->name);
++
+ 	ast_module_ref(ast_module_info->self);
+ 	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
+ 	/*Since it is valid to have extensions in the dialplan that have unescaped characters in them
+@@ -4479,8 +4485,10 @@
+ 	build_via(p);
+ 	if (!callid)
+ 		build_callid_pvt(p);
+-	else
++	else {
+ 		ast_string_field_set(p, callid, callid);
++		ast_rtp_set_chan_id(p->rtp, p->callid);
++	}
+ 	/* Assign default music on hold class */
+ 	ast_string_field_set(p, mohinterpret, default_mohinterpret);
+ 	ast_string_field_set(p, mohsuggest, default_mohsuggest);
+--- include/asterisk/rtp.h.orig	2008-03-18 13:35:42.000000000 +0200
++++ include/asterisk/rtp.h	2008-03-18 13:35:58.000000000 +0200
+@@ -251,6 +251,9 @@
+ 
+ int ast_rtp_codec_getformat(int pt);
+ 
++void ast_rtp_set_chan_name(struct ast_rtp *, const char *);
++void ast_rtp_set_chan_id(struct ast_rtp *, const char *);
++
+ /*! \brief Set rtp timeout */
+ void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+ /*! \brief Set rtp hold timeout */
+--- main/rtp.c.orig	2008-04-08 14:53:18.000000000 +0300
++++ main/rtp.c	2008-04-08 14:54:14.000000000 +0300
+@@ -81,6 +81,7 @@
+ static int rtpstart;			/*!< First port for RTP sessions (set in rtp.conf) */
+ static int rtpend;			/*!< Last port for RTP sessions (set in rtp.conf) */
+ static int rtpdebug;			/*!< Are we debugging? */
++static int rtpdebugdtmf;		/*!< Are we debugging DTMFs? */
+ static int rtcpdebug;			/*!< Are we debugging RTCP? */
+ static int rtcpstats;			/*!< Are we debugging RTCP? */
+ static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
+@@ -168,6 +169,8 @@
+ 	struct ast_codec_pref pref;
+ 	struct ast_rtp *bridged;        /*!< Who we are Packet bridged to */
+ 	int set_marker_bit:1;           /*!< Whether to set the marker bit or not */
++	char chan_name[100];
++	char chan_id[100];
+ };
+ 
+ /* Forward declarations */
+@@ -669,8 +672,8 @@
+ 	struct ast_frame *f = NULL;
+ 	event = ntohl(*((unsigned int *)(data)));
+ 	event &= 0x001F;
+-	if (option_debug > 2 || rtpdebug)
+-		ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
++	if (option_debug > 2 || rtpdebug || rtpdebugdtmf)
++		ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF packet: %08x (len = %d)\n", rtp->chan_name, rtp->chan_id, event, len);
+ 	if (event < 10) {
+ 		resp = '0' + event;
+ 	} else if (event < 11) {
+@@ -684,12 +687,24 @@
+ 	}
+ 	if (rtp->resp && (rtp->resp != resp)) {
+ 		f = send_dtmf(rtp, AST_FRAME_DTMF_END);
++		ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF event: %c\n", rtp->chan_name, rtp->chan_id, rtp->resp);
+ 	}
+ 	rtp->resp = resp;
+ 	rtp->dtmfcount = dtmftimeout;
+ 	return f;
+ }
+ 
++void ast_rtp_set_chan_id(struct ast_rtp *rtp, const char *chan_id) {
++	if (rtp == NULL || chan_id == NULL)
++		return;
++	snprintf(rtp->chan_id, sizeof(rtp->chan_id), "%s", chan_id);
++}
++
++void ast_rtp_set_chan_name(struct ast_rtp *rtp, const char *chan_name) {
++	if (rtp == NULL || chan_name == NULL)
++		return;
++	snprintf(rtp->chan_name, sizeof(rtp->chan_name), "%s", chan_name);
++}
+ /*! 
+  * \brief Process RTP DTMF and events according to RFC 2833.
+  * 
+@@ -1051,6 +1066,10 @@
+ 	struct rtpPayloadType rtpPT;
+ 	int reconstruct = ntohl(rtpheader[0]);
+ 
++	/* If we are listening for DTMF - then feed all packets into the core  to keep the RTP stream consistent when relaying DTMFs */
++	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF))
++		return -1;
++
+ 	/* Get fields from packet */
+ 	payload = (reconstruct & 0x7f0000) >> 16;
+ 	mark = (((reconstruct & 0x800000) >> 23) != 0);
+@@ -1062,10 +1081,6 @@
+ 	if (!bridged->current_RTP_PT[payload].code)
+ 		return -1;
+ 
+-	/* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
+-	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
+-		return -1;
+-
+ 	/* Otherwise adjust bridged payload to match */
+ 	bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
+ 
+@@ -1254,11 +1269,12 @@
+ 		/* This is special in-band data that's not one of our codecs */
+ 		if (rtpPT.code == AST_RTP_DTMF) {
+ 			/* It's special -- rfc2833 process it */
+-			if (rtp_debug_test_addr(&sin)) {
++			if (rtp_debug_test_addr(&sin) || rtpdebugdtmf) {
+ 				unsigned char *data;
+ 				unsigned int event;
+ 				unsigned int event_end;
+ 				unsigned int duration;
++
+ 				data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
+ 				event = ntohl(*((unsigned int *)(data)));
+ 				event >>= 24;
+@@ -1267,9 +1283,12 @@
+ 				event_end >>= 24;
+ 				duration = ntohl(*((unsigned int *)(data)));
+ 				duration &= 0xFFFF;
+-				ast_verbose("Got  RTP RFC2833 from   %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
++
++				ast_verbose("Channel: %s %s Got RTP RFC2833 from %s:%u to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d)\n", rtp->chan_name, rtp->chan_id, ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->us.sin_addr), ntohs(rtp->us.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
+ 			}
+ 			f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
++			if (rtpdebugdtmf && f)
++				ast_verbose("Channel: %s %s Got RFC2833 DTMF event %c of type %s\n", rtp->chan_name, rtp->chan_id, f->subclass, (f->frametype == AST_FRAME_DTMF_BEGIN ? "DTMF BEGIN" : (f->frametype == AST_FRAME_DTMF_END ? "DTMF_END" : "UNKNOWN")));
+ 		} else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
+ 			/* It's really special -- process it the Cisco way */
+ 			if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
+@@ -2198,8 +2217,9 @@
+ 			ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
+ 				ast_inet_ntoa(rtp->them.sin_addr),
+ 				ntohs(rtp->them.sin_port), strerror(errno));
+-		if (rtp_debug_test_addr(&rtp->them))
+-			ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++		if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++			ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++				    rtp->chan_name, rtp->chan_id,
+ 				    ast_inet_ntoa(rtp->them.sin_addr),
+ 				    ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ 		/* Increment sequence number */
+@@ -2242,8 +2262,9 @@
+ 		ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ 			ast_inet_ntoa(rtp->them.sin_addr),
+ 			ntohs(rtp->them.sin_port), strerror(errno));
+-	if (rtp_debug_test_addr(&rtp->them))
+-		ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++	if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++		ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++			    rtp->chan_name, rtp->chan_id,
+ 			    ast_inet_ntoa(rtp->them.sin_addr),
+ 			    ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ 
+@@ -3481,6 +3502,16 @@
+ 	return RESULT_SUCCESS;
+ }
+ 
++static int rtp_do_debug_dtmf(int fd, int argc, char *argv[])
++{
++	if (argc != 3)
++		return RESULT_SHOWUSAGE;
++
++	rtpdebugdtmf = 1;
++	ast_cli(fd, "RTP DTMF debugging enabled\n");
++	return RESULT_SUCCESS;
++}
++
+ static int rtp_do_debug(int fd, int argc, char *argv[])
+ {
+ 	if (argc != 2) {
+@@ -3541,6 +3572,7 @@
+ 	if (argc != 3)
+ 		return RESULT_SHOWUSAGE;
+ 	rtpdebug = 0;
++	rtpdebugdtmf = 0;
+ 	ast_cli(fd,"RTP Debugging Disabled\n");
+ 	return RESULT_SUCCESS;
+ }
+@@ -3601,7 +3633,7 @@
+ }
+ 
+ static char debug_usage[] =
+-  "Usage: rtp debug [ip host[:port]]\n"
++  "Usage: rtp debug [ip host[:port] | dtmf]\n"
+   "       Enable dumping of all RTP packets to and from host.\n";
+ 
+ static char no_debug_usage[] =
+@@ -3676,6 +3708,10 @@
+ 	rtp_do_debug, "Enable RTP debugging",
+ 	debug_usage },
+ 
++	{ { "rtp", "debug", "dtmf", NULL },
++	rtp_do_debug_dtmf, "Enable RTP debugging on DTMFs",
++	debug_usage },
++
+ 	{ { "rtp", "debug", "off", NULL },
+ 	rtp_no_debug, "Disable RTP debugging",
+ 	no_debug_usage, NULL, &cli_rtp_no_debug_deprecated },