Merged revisions 230881 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
  
  Change fax detection in chan_sip so it behaves as one would expect.
  
  Internally the way T.38 is negotiated has changed and the option no longer
  reflects a behavior that is valid. It will now look for a CNG tone on
  received calls and if present send the call to the 'fax' extension. It is
  then up to the application or channel to request the switch over to T.38.
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230883 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 61 insertions, 35 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 54e449f90..1ee0c5047 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1091,6 +1091,7 @@ struct sip_auth {
 #define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)	/*!< DP: Compensate for buggy RFC2833 implementations */
 #define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< DP: Buggy CISCO MWI fix */
 #define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 27)       /*!< 29: Has a dialog been established? */
+#define SIP_PAGE2_FAX_DETECT		(1 << 28)		/*!< DP: Fax Detection support */
 #define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
 #define SIP_PAGE2_VIDEOSUPPORT_ALWAYS	(1 << 31)       /*!< DP: Always set up video, even if endpoints don't support it */
@@ -1099,7 +1100,7 @@ struct sip_auth {
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
 	SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
 	SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | \
-	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC)
+	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC | SIP_PAGE2_FAX_DETECT)
 
 /*@}*/ 
 
@@ -1364,7 +1365,7 @@ struct sip_pvt {
 	int laststate;				/*!< SUBSCRIBE: Last known extension state */
 	int dialogver;				/*!< SUBSCRIBE: Version for subscription dialog-info */
 
-	struct ast_dsp *vad;			/*!< Inband DTMF Detection dsp */
+	struct ast_dsp *dsp;			/*!< A DSP for inband DTMF and fax CNG tone detection */
 
 	struct sip_peer *relatedpeer;		/*!< If this dialog is related to a peer, which one 
 							Used in peerpoke, mwi subscriptions */
@@ -5607,8 +5608,8 @@ static int sip_hangup(struct ast_channel *ast)
 	append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
 
 	/* Disconnect */
-	if (p->vad)
-		ast_dsp_free(p->vad);
+	if (p->dsp)
+		ast_dsp_free(p->dsp);
 
 	p->owner = NULL;
 	ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");
@@ -6250,12 +6251,19 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	else
 		ast_debug(3, "This channel will not be able to handle video.\n");
 
-	if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
-		i->vad = ast_dsp_new();
-		ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
+	if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) ||
+	    (ast_test_flag(&i->flags[1], SIP_PAGE2_FAX_DETECT))) {
+		int features = DSP_FEATURE_DIGIT_DETECT;
+
+		if (ast_test_flag(&i->flags[1], SIP_PAGE2_FAX_DETECT)) {
+			features |= DSP_FEATURE_FAX_DETECT;
+		}
+
+		i->dsp = ast_dsp_new();
+		ast_dsp_set_features(i->dsp, features);
 		if (global_relaxdtmf)
-			ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
-	}
+			ast_dsp_set_digitmode(i->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
+         }
 
 	/* Set file descriptors for audio, video, realtime text and UDPTL as needed */
 	if (i->rtp) {
@@ -6566,12 +6574,20 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
 		ast_set_write_format(p->owner, p->owner->writeformat);
 	}
 
-	if (f && (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
-		f = ast_dsp_process(p->owner, p->vad, f);
-		if (f && f->frametype == AST_FRAME_DTMF) {
-			if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && f->subclass == 'f') {
-				ast_debug(1, "Fax CNG detected on %s\n", ast->name);
-				*faxdetect = 1;
+	if (f && ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT)) && p->dsp) {
+		f = ast_dsp_process(p->owner, p->dsp, f);
+                if (f && f->frametype == AST_FRAME_DTMF) {
+			if (f->subclass == 'f') {
+                                if (option_debug)
+                                        ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+                                *faxdetect = 1;
+				/* If we only needed this DSP for fax detection purposes we can just drop it now */
+				if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
+					ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
+				} else {
+					ast_dsp_free(p->dsp);
+					p->dsp = NULL;
+				}
 			} else {
 				ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
 			}
@@ -6592,20 +6608,26 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 	fr = sip_rtp_read(ast, p, &faxdetected);
 	p->lastrtprx = time(NULL);
 
-	/* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
-	/* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
-	if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
-		if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
-			if (!p->pendinginvite) {
-				ast_debug(3, "Sending reinvite on SIP (%s) for T.38 negotiation.\n", ast->name);
-				change_t38_state(p, T38_LOCAL_REINVITE);
-				transmit_reinvite_with_sdp(p, TRUE, FALSE);
-			}
-		} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
-			ast_debug(3, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
-			ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
-		}
-	}
+	/* If we detect a CNG tone and fax detection is enabled then send us off to the fax extension */
+	if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT)) {
+		ast_channel_lock(ast);
+		if (strcmp(ast->exten, "fax")) {
+			const char *target_context = S_OR(ast->macrocontext, ast->context);
+			ast_channel_unlock(ast);
+			if (ast_exists_extension(ast, target_context, "fax", 1, ast->cid.cid_num)) {
+				ast_verbose(VERBOSE_PREFIX_2 "Redirecting '%s' to fax extension\n", ast->name);
+				pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
+				if (ast_async_goto(ast, target_context, "fax", 1)) {
+					ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
+				}
+				fr = &ast_null_frame;
+			} else {
+				ast_log(LOG_NOTICE, "Fax detected but no fax extension\n");
+                        }
+		} else {
+			ast_channel_unlock(ast);
+                }
+        }
 
 	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
 	if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
@@ -22330,6 +22352,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "constantssrc")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
+	} else if (!strcasecmp(v->name, "faxdetect")) {
+                ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
+                ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FAX_DETECT);
 	} else
 		res = 0;
 
@@ -23358,6 +23383,7 @@ static int reload_config(enum channelreloadreason reason)
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
+	ast_clear_flag(&global_flags[1], SIP_PAGE2_FAX_DETECT);
 
 
 	/* Read the [general] config section of sip.conf (or from realtime config) */
@@ -24251,14 +24277,14 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
 	if (p->rtp)
 		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 	if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
-		if (!p->vad) {
-			p->vad = ast_dsp_new();
-			ast_dsp_set_features(p->vad, DSP_FEATURE_DIGIT_DETECT);
+		if (!p->dsp) {
+			p->dsp = ast_dsp_new();
+			ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
 		}
 	} else {
-		if (p->vad) {
-			ast_dsp_free(p->vad);
-			p->vad = NULL;
+		if (p->dsp) {
+			ast_dsp_free(p->dsp);
+			p->dsp = NULL;
 		}
 	}
 	sip_pvt_unlock(p);