[asterisk-commits] lmadsen: tag 1.6.2.17-rc2 r303141 - in /tags/1.6.2.17-rc2: ./ main/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Jan 20 14:24:59 CST 2011


Author: lmadsen
Date: Thu Jan 20 14:24:53 2011
New Revision: 303141

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=303141
Log:
Update .version, ChangeLog, and merge changes.

Removed:
    tags/1.6.2.17-rc2/asterisk-1.6.2.17-rc1-summary.html
    tags/1.6.2.17-rc2/asterisk-1.6.2.17-rc1-summary.txt
Modified:
    tags/1.6.2.17-rc2/   (props changed)
    tags/1.6.2.17-rc2/.version
    tags/1.6.2.17-rc2/ChangeLog
    tags/1.6.2.17-rc2/main/features.c

Propchange: tags/1.6.2.17-rc2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: tags/1.6.2.17-rc2/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.17-rc2/.version?view=diff&rev=303141&r1=303140&r2=303141
==============================================================================
--- tags/1.6.2.17-rc2/.version (original)
+++ tags/1.6.2.17-rc2/.version Thu Jan 20 14:24:53 2011
@@ -1,1 +1,1 @@
-1.6.2.17-rc1
+1.6.2.17-rc2

Modified: tags/1.6.2.17-rc2/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.17-rc2/ChangeLog?view=diff&rev=303141&r1=303140&r2=303141
==============================================================================
--- tags/1.6.2.17-rc2/ChangeLog (original)
+++ tags/1.6.2.17-rc2/ChangeLog Thu Jan 20 14:24:53 2011
@@ -1,3 +1,136 @@
+2011-01-20  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.17-rc2 Released.
+
+	------------------------------------------------------------------------
+	r302172 | rmudgett | 2011-01-18 12:04:37 -0600 (Tue, 18 Jan 2011) | 88
+	lines
+
+	Issues with DTMF triggered attended transfers.
+
+	Issue 0017999
+	1) A calls B. B answers.
+	2) B using DTMF dial *2 (code in features.conf for attended transfer).
+	3) A hears MOH. B dial number C
+	4) C ringing. A hears MOH.
+	5) B hangup. A still hears MOH. C ringing.
+	6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
+	For v1.4 C will ring forever until C answers the dead line. (Issue
+	0017096)
+
+	Problem: When A and B hangup, C is still ringing.
+
+	Issue 0018395
+	SIP call limit of B is 1
+	1. A call B, B answered
+	2. B *2(atxfer) call C
+	3. B hangup, C ringing
+	4. Timeout waiting for C to answer
+	5. Recall to B fails because B has reached its call limit.
+
+	Because B reached its call limit, it cannot do anything until the
+	transfer
+	it started completes.
+
+	Issue 0017273
+	Same scenario as issue 18395 but party B is an FXS port. Party B
+	cannot
+	do anything until the transfer it started completes. If B goes back
+	off
+	hook before C answers, B hears ringback instead of the expected
+	dialtone.
+
+	**********
+	Note for the issue 0017273 and 0018395 fix:
+
+	DTMF attended transfer works within the channel bridge. Unfortunately,
+	when either party A or B in the channel bridge hangs up, that channel
+	is
+	not completely hung up until the transfer completes. This is a real
+	problem depending upon the channel technology involved.
+
+	For chan_dahdi, the channel is crippled until the hangup is complete.
+	Either the channel is not useable (analog) or the protocol disconnect
+	messages are held up (PRI/BRI/SS7) and the media is not released.
+
+	For chan_sip, a call limit of one is going to block that endpoint from
+	any
+	further calls until the hangup is complete.
+
+	For party A this is a minor problem. The party A channel will only be
+	in
+	this condition while party B is dialing and when party B and C are
+	conferring. The conversation between party B and C is expected to be a
+	short one. Party B is either asking a question of party C or
+	announcing
+	party A. Also party A does not have much incentive to hangup at this
+	point.
+
+	For party B this can be a major problem during a blonde transfer. (A
+	blonde transfer is our term for an attended transfer that is converted
+	into a blind transfer. :)) Party B could be the operator. When party B
+	hangs up, he assumes that he is out of the original call entirely. The
+	party B channel will be in this condition while party C is ringing,
+	while
+	attempting to recall party B, and while waiting between call attempts.
+
+	WARNING:
+	The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will
+	replace the party B channel technology with a NULL channel driver to
+	complete hanging up the party B channel technology. The consequences
+	of
+	this code is that the 'h' extension will not be able to access any
+	channel
+	technology specific information like SIP statistics for the call.
+
+	ATXFER_NULL_TECH is not defined by default.
+	**********
+
+	(closes issue 0017999)
+	Reported by: iskatel
+	Tested by: rmudgett
+	JIRA SWP-2246
+
+	(closes issue 0017096)
+	Reported by: gelo
+	Tested by: rmudgett
+	JIRA SWP-1192
+
+	(closes issue 0018395)
+	Reported by: shihchuan
+	Tested by: rmudgett
+
+	(closes issue 0017273)
+	Reported by: grecco
+	Tested by: rmudgett
+
+	Review: https://reviewboard.asterisk.org/r/1047/ [^]
+
+	------------------------------------------------------------------------
+
+	------------------------------------------------------------------------
+	r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
+	lines
+
+	main/features: Use POLLPRI when waiting for events on parked channels.
+
+	This change resolves a regression in the 1.6.2 when converting from
+	select to poll. The DAHDI timers use POLLPRI to indicate that the
+	timer
+	fired, but features was not waiting for that flag. The result was no
+	audio for MOH when a call was parked and res_timing_dahdi was in use.
+
+	This patch is slightly modified from the one on the mantis issue. It
+	does
+	not set an exception on the channel if the POLLPRI flag is set.
+
+	(closes issue 0018262)
+	Reported by: francesco_r
+	Patches:
+	      patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
+	            Tested by: francesco_r, rfrantik, one47
+	------------------------------------------------------------------------
+
 2011-01-14  Leif Madsen <lmadsen at digium.com>
 
 	* Asterisk 1.6.2.17-rc1 Released.

Modified: tags/1.6.2.17-rc2/main/features.c
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.17-rc2/main/features.c?view=diff&rev=303141&r1=303140&r2=303141
==============================================================================
--- tags/1.6.2.17-rc2/main/features.c (original)
+++ tags/1.6.2.17-rc2/main/features.c Thu Jan 20 14:24:53 2011
@@ -56,6 +56,56 @@
 #include "asterisk/audiohook.h"
 #include "asterisk/global_datastores.h"
 #include "asterisk/astobj2.h"
+
+/*
+ * Party A - transferee
+ * Party B - transferer
+ * Party C - target of transfer
+ *
+ * DTMF attended transfer works within the channel bridge.
+ * Unfortunately, when either party A or B in the channel bridge
+ * hangs up, that channel is not completely hung up until the
+ * transfer completes.  This is a real problem depending upon
+ * the channel technology involved.
+ *
+ * For chan_dahdi, the channel is crippled until the hangup is
+ * complete.  Either the channel is not useable (analog) or the
+ * protocol disconnect messages are held up (PRI/BRI/SS7) and
+ * the media is not released.
+ *
+ * For chan_sip, a call limit of one is going to block that
+ * endpoint from any further calls until the hangup is complete.
+ *
+ * For party A this is a minor problem.  The party A channel
+ * will only be in this condition while party B is dialing and
+ * when party B and C are conferring.  The conversation between
+ * party B and C is expected to be a short one.  Party B is
+ * either asking a question of party C or announcing party A.
+ * Also party A does not have much incentive to hangup at this
+ * point.
+ *
+ * For party B this can be a major problem during a blonde
+ * transfer.  (A blonde transfer is our term for an attended
+ * transfer that is converted into a blind transfer. :))  Party
+ * B could be the operator.  When party B hangs up, he assumes
+ * that he is out of the original call entirely.  The party B
+ * channel will be in this condition while party C is ringing,
+ * while attempting to recall party B, and while waiting between
+ * call attempts.
+ *
+ * WARNING:
+ * The ATXFER_NULL_TECH conditional is a hack to fix the
+ * problem.  It will replace the party B channel technology with
+ * a NULL channel driver.  The consequences of this code is that
+ * the 'h' extension will not be able to access any channel
+ * technology specific information like SIP statistics for the
+ * call.
+ *
+ * Uncomment the ATXFER_NULL_TECH define below to replace the
+ * party B channel technology in the channel bridge to complete
+ * hanging up the channel technology.
+ */
+//#define ATXFER_NULL_TECH	1
 
 /*** DOCUMENTATION
 	<application name="Bridge" language="en_US">
@@ -278,6 +328,119 @@
 	int is_caller;
 };
 
+#if defined(ATXFER_NULL_TECH)
+static struct ast_frame *null_read(struct ast_channel *chan)
+{
+	/* Hangup channel. */
+	return NULL;
+}
+
+static struct ast_frame *null_exception(struct ast_channel *chan)
+{
+	/* Hangup channel. */
+	return NULL;
+}
+
+static int null_write(struct ast_channel *chan, struct ast_frame *frame)
+{
+	/* Hangup channel. */
+	return -1;
+}
+
+static int null_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+	/* No problem fixing up the channel. */
+	return 0;
+}
+
+static int null_hangup(struct ast_channel *chan)
+{
+	chan->tech_pvt = NULL;
+	return 0;
+}
+
+static const struct ast_channel_tech null_tech = {
+	.type = "NULL",
+	.description = "NULL channel driver for atxfer",
+	.capabilities = -1,
+	.read = null_read,
+	.exception = null_exception,
+	.write = null_write,
+	.fixup = null_fixup,
+	.hangup = null_hangup,
+};
+#endif	/* defined(ATXFER_NULL_TECH) */
+
+#if defined(ATXFER_NULL_TECH)
+/*!
+ * \internal
+ * \brief Set the channel technology to the NULL technology.
+ *
+ * \param chan Channel to change technology.
+ *
+ * \return Nothing
+ */
+static void set_null_chan_tech(struct ast_channel *chan)
+{
+	int idx;
+
+	ast_channel_lock(chan);
+
+	/* Hangup the channel's physical side */
+	if (chan->tech->hangup) {
+		chan->tech->hangup(chan);
+	}
+	if (chan->tech_pvt) {
+		ast_log(LOG_WARNING, "Channel '%s' may not have been hung up properly\n",
+			chan->name);
+		ast_free(chan->tech_pvt);
+		chan->tech_pvt = NULL;
+	}
+
+	/* Install the NULL technology and wake up anyone waiting on it. */
+	chan->tech = &null_tech;
+	for (idx = 0; idx < AST_MAX_FDS; ++idx) {
+		switch (idx) {
+		case AST_ALERT_FD:
+		case AST_TIMING_FD:
+		case AST_GENERATOR_FD:
+			/* Don't clear these fd's. */
+			break;
+		default:
+			ast_channel_set_fd(chan, idx, -1);
+			break;
+		}
+	}
+	ast_queue_frame(chan, &ast_null_frame);
+
+	ast_channel_unlock(chan);
+}
+#endif	/* defined(ATXFER_NULL_TECH) */
+
+#if defined(ATXFER_NULL_TECH)
+/*!
+ * \internal
+ * \brief Set the channel name to something unique.
+ *
+ * \param chan Channel to change name.
+ *
+ * \return Nothing
+ */
+static void set_new_chan_name(struct ast_channel *chan)
+{
+	char *orig_name;
+	static int seq_num;
+
+	ast_channel_lock(chan);
+
+	orig_name = ast_strdupa(chan->name);
+	ast_string_field_build(chan, name, "%s<XFER_%x>", orig_name,
+		ast_atomic_fetchadd_int(&seq_num, +1));
+
+	ast_channel_unlock(chan);
+}
+#endif	/* defined(ATXFER_NULL_TECH) */
+
 static void *dial_features_duplicate(void *data)
 {
 	struct ast_dial_features *df = data, *df_copy;
@@ -400,7 +563,10 @@
 	}
 }
 
-static struct ast_channel *feature_request_and_dial(struct ast_channel *caller, struct ast_channel *transferee, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, int igncallerstate, const char *language);
+static struct ast_channel *feature_request_and_dial(struct ast_channel *caller,
+	const char *caller_name, struct ast_channel *transferee, const char *type,
+	int format, void *data, int timeout, int *outstate, const char *cid_num,
+	const char *cid_name, const char *language);
 
 /*!
  * \brief bridge the call 
@@ -1392,9 +1558,9 @@
  * \param config
  * \param code
  * \param sense feature options
- * 
+ *
  * \param data
- * Get extension to transfer to, if you cannot generate channel (or find extension) 
+ * Get extension to transfer to, if you cannot generate channel (or find extension)
  * return to host channel. After called channel answered wait for hangup of transferer,
  * bridge call between transfer peer (taking them off hold) to attended transfer channel.
  *
@@ -1402,8 +1568,8 @@
 */
 static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data)
 {
-	struct ast_channel *transferer;
-	struct ast_channel *transferee;
+	struct ast_channel *transferer;/* Party B */
+	struct ast_channel *transferee;/* Party A */
 	const char *transferer_real_context;
 	char xferto[256] = "";
 	int res;
@@ -1412,60 +1578,60 @@
 	struct ast_channel *xferchan;
 	struct ast_bridge_thread_obj *tobj;
 	struct ast_bridge_config bconfig;
-	struct ast_frame *f;
 	int l;
 	struct ast_datastore *features_datastore;
 	struct ast_dial_features *dialfeatures = NULL;
+	char *transferer_tech;
+	char *transferer_name;
+	char *transferer_name_orig;
+	char *dash;
 
 	ast_debug(1, "Executing Attended Transfer %s, %s (sense=%d) \n", chan->name, peer->name, sense);
 	set_peers(&transferer, &transferee, peer, chan, sense);
 	transferer_real_context = real_ctx(transferer, transferee);
-	/* Start autoservice on chan while we talk to the originator */
+
+	/* Start autoservice on transferee while we talk to the transferer */
 	ast_autoservice_start(transferee);
-	ast_autoservice_ignore(transferee, AST_FRAME_DTMF_END);
 	ast_indicate(transferee, AST_CONTROL_HOLD);
-	
+
 	/* Transfer */
 	res = ast_stream_and_wait(transferer, "pbx-transfer", AST_DIGIT_ANY);
 	if (res < 0) {
 		finishup(transferee);
-		return res;
+		return -1;
 	}
 	if (res > 0) /* If they've typed a digit already, handle it */
 		xferto[0] = (char) res;
 
 	/* this is specific of atxfer */
 	res = ast_app_dtget(transferer, transferer_real_context, xferto, sizeof(xferto), 100, transferdigittimeout);
-	if (res < 0) {  /* hangup, would be 0 for invalid and 1 for valid */
+	if (res < 0) {  /* hangup or error, (would be 0 for invalid and 1 for valid) */
 		finishup(transferee);
-		return res;
-	}
+		return -1;
+	}
+	l = strlen(xferto);
 	if (res == 0) {
-		ast_log(LOG_WARNING, "Did not read data.\n");
+		if (l) {
+			ast_log(LOG_WARNING, "Extension '%s' does not exist in context '%s'\n",
+				xferto, transferer_real_context);
+		} else {
+			/* Does anyone care about this case? */
+			ast_log(LOG_WARNING, "No digits dialed for atxfer.\n");
+		}
+		ast_stream_and_wait(transferer, "pbx-invalid", "");
 		finishup(transferee);
-		if (ast_stream_and_wait(transferer, "beeperr", ""))
-			return -1;
 		return AST_FEATURE_RETURN_SUCCESS;
 	}
 
-	/* valid extension, res == 1 */
-	if (!ast_exists_extension(transferer, transferer_real_context, xferto, 1, transferer->cid.cid_num)) {
-		ast_log(LOG_WARNING, "Extension %s does not exist in context %s\n",xferto,transferer_real_context);
+	/* If we are attended transfering to parking, just use builtin_parkcall instead of trying to track all of
+	 * the different variables for handling this properly with a builtin_atxfer */
+	if (!strcmp(xferto, ast_parking_ext())) {
 		finishup(transferee);
-		if (ast_stream_and_wait(transferer, "beeperr", ""))
-			return -1;
-		return AST_FEATURE_RETURN_SUCCESS;
-	}
-
- 	/* If we are attended transfering to parking, just use builtin_parkcall instead of trying to track all of
- 	 * the different variables for handling this properly with a builtin_atxfer */
- 	if (!strcmp(xferto, ast_parking_ext())) {
- 		finishup(transferee);
- 		return builtin_parkcall(chan, peer, config, code, sense, data);
- 	}
-
-	l = strlen(xferto);
-	snprintf(xferto + l, sizeof(xferto) - l, "@%s/n", transferer_real_context);	/* append context */
+		return builtin_parkcall(chan, peer, config, code, sense, data);
+	}
+
+	/* Append context to dialed transfer number. */
+	snprintf(xferto + l, sizeof(xferto) - l, "@%s/n", transferer_real_context);
 
 	/* If we are performing an attended transfer and we have two channels involved then
 	   copy sound file information to play upon attended transfer completion */
@@ -1481,29 +1647,66 @@
 		}
 	}
 
-	newchan = feature_request_and_dial(transferer, transferee, "Local", ast_best_codec(transferer->nativeformats),
-		xferto, atxfernoanswertimeout, &outstate, transferer->cid.cid_num, transferer->cid.cid_name, 1, transferer->language);
+	/* Extract redial transferer information from the channel name. */
+	transferer_name_orig = ast_strdupa(transferer->name);
+	transferer_name = ast_strdupa(transferer_name_orig);
+	transferer_tech = strsep(&transferer_name, "/");
+	dash = strrchr(transferer_name, '-');
+	if (dash) {
+		/* Trim off channel name sequence/serial number. */
+		*dash = '\0';
+	}
+
+	/* Stop autoservice so we can monitor all parties involved in the transfer. */
+	if (ast_autoservice_stop(transferee) < 0) {
+		ast_indicate(transferee, AST_CONTROL_UNHOLD);
+		return -1;
+	}
+
+	/* Dial party C */
+	newchan = feature_request_and_dial(transferer, transferer_name_orig, transferee,
+		"Local", ast_best_codec(transferer->nativeformats), xferto, atxfernoanswertimeout,
+		&outstate, transferer->cid.cid_num, transferer->cid.cid_name,
+		transferer->language);
+	ast_debug(2, "Dial party C result: newchan:%d, outstate:%d\n", !!newchan, outstate);
 
 	if (!ast_check_hangup(transferer)) {
 		int hangup_dont = 0;
 
-		/* Transferer is up - old behaviour */
+		/* Transferer (party B) is up */
+		ast_debug(1, "Actually doing an attended transfer.\n");
+
+		/* Start autoservice on transferee while the transferer deals with party C. */
+		ast_autoservice_start(transferee);
+
 		ast_indicate(transferer, -1);
 		if (!newchan) {
+			/* any reason besides user requested cancel and busy triggers the failed sound */
+			switch (outstate) {
+			case AST_CONTROL_UNHOLD:/* Caller requested cancel or party C answer timeout. */
+			case AST_CONTROL_BUSY:
+			case AST_CONTROL_CONGESTION:
+				if (ast_stream_and_wait(transferer, xfersound, "")) {
+					ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
+				}
+				break;
+			default:
+				if (ast_stream_and_wait(transferer, xferfailsound, "")) {
+					ast_log(LOG_WARNING, "Failed to play transfer failed sound!\n");
+				}
+				break;
+			}
 			finishup(transferee);
-			/* any reason besides user requested cancel and busy triggers the failed sound */
-			if (outstate != AST_CONTROL_UNHOLD && outstate != AST_CONTROL_BUSY &&
-				ast_stream_and_wait(transferer, xferfailsound, ""))
-				return -1;
-			if (ast_stream_and_wait(transferer, xfersound, ""))
-				ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
 			return AST_FEATURE_RETURN_SUCCESS;
 		}
 
 		if (check_compat(transferer, newchan)) {
+			if (ast_stream_and_wait(transferer, xferfailsound, "")) {
+				ast_log(LOG_WARNING, "Failed to play transfer failed sound!\n");
+			}
 			/* we do mean transferee here, NOT transferer */
 			finishup(transferee);
-			return -1;
+			return AST_FEATURE_RETURN_SUCCESS;
 		}
 		memset(&bconfig,0,sizeof(struct ast_bridge_config));
 		ast_set_flag(&(bconfig.features_caller), AST_FEATURE_DISCONNECT);
@@ -1515,205 +1718,194 @@
 		if (ast_test_flag(chan, AST_FLAG_BRIDGE_HANGUP_DONT)) {
 			hangup_dont = 1;
 		}
-		res = ast_bridge_call(transferer, newchan, &bconfig);
+		/* Let party B and party C talk as long as they want. */
+		ast_bridge_call(transferer, newchan, &bconfig);
 		if (hangup_dont) {
 			ast_set_flag(chan, AST_FLAG_BRIDGE_HANGUP_DONT);
 		}
 
 		if (ast_check_hangup(newchan) || !ast_check_hangup(transferer)) {
 			ast_hangup(newchan);
-			if (ast_stream_and_wait(transferer, xfersound, ""))
+			if (ast_stream_and_wait(transferer, xfersound, "")) {
 				ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
+			}
 			finishup(transferee);
-			transferer->_softhangup = 0;
 			return AST_FEATURE_RETURN_SUCCESS;
 		}
+
+		/* Transferer (party B) is confirmed hung up at this point. */
 		if (check_compat(transferee, newchan)) {
 			finishup(transferee);
 			return -1;
 		}
+
 		ast_indicate(transferee, AST_CONTROL_UNHOLD);
-
 		if ((ast_autoservice_stop(transferee) < 0)
-		 || (ast_waitfordigit(transferee, 100) < 0)
-		 || (ast_waitfordigit(newchan, 100) < 0)
-		 || ast_check_hangup(transferee)
-		 || ast_check_hangup(newchan)) {
+			|| (ast_waitfordigit(transferee, 100) < 0)
+			|| (ast_waitfordigit(newchan, 100) < 0)
+			|| ast_check_hangup(transferee)
+			|| ast_check_hangup(newchan)) {
 			ast_hangup(newchan);
 			return -1;
 		}
-		xferchan = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, "Transfered/%s", transferee->name);
-		if (!xferchan) {
+	} else if (!ast_check_hangup(transferee)) {
+		/* Transferer (party B) has hung up at this point.  Doing blonde transfer. */
+		ast_debug(1, "Actually doing a blonde transfer.\n");
+
+		if (!newchan && !atxferdropcall) {
+			/* Party C is not available, try to call party B back. */
+			unsigned int tries = 0;
+
+			if (ast_strlen_zero(transferer_name) || ast_strlen_zero(transferer_tech)) {
+				ast_log(LOG_WARNING,
+					"Transferer channel name: '%s' cannot be used for callback.\n",
+					transferer_name_orig);
+				ast_indicate(transferee, AST_CONTROL_UNHOLD);
+				return -1;
+			}
+
+			tries = 0;
+			for (;;) {
+				/* Try to get party B back. */
+				ast_debug(1, "We're trying to callback %s/%s\n",
+					transferer_tech, transferer_name);
+				newchan = feature_request_and_dial(transferer, transferer_name_orig,
+					transferee, transferer_tech,
+					ast_best_codec(transferee->nativeformats), transferer_name,
+					atxfernoanswertimeout, &outstate, transferee->cid.cid_num,
+					transferee->cid.cid_name, transferer->language);
+				ast_debug(2, "Dial party B result: newchan:%d, outstate:%d\n",
+					!!newchan, outstate);
+				if (newchan || ast_check_hangup(transferee)) {
+					break;
+				}
+
+				++tries;
+				if (atxfercallbackretries <= tries) {
+					/* No more callback tries remaining. */
+					break;
+				}
+
+				if (atxferloopdelay) {
+					/* Transfer failed, sleeping */
+					ast_debug(1, "Sleeping for %d ms before retrying atxfer.\n",
+						atxferloopdelay);
+					ast_safe_sleep(transferee, atxferloopdelay);
+					if (ast_check_hangup(transferee)) {
+						return -1;
+					}
+				}
+
+				/* Retry dialing party C. */
+				ast_debug(1, "We're retrying to call %s/%s\n", "Local", xferto);
+				newchan = feature_request_and_dial(transferer, transferer_name_orig,
+					transferee, "Local", ast_best_codec(transferee->nativeformats),
+					xferto, atxfernoanswertimeout, &outstate, transferer->cid.cid_num,
+					transferer->cid.cid_name, transferer->language);
+				ast_debug(2, "Redial party C result: newchan:%d, outstate:%d\n",
+					!!newchan, outstate);
+				if (newchan || ast_check_hangup(transferee)) {
+					break;
+				}
+			}
+		}
+		ast_indicate(transferee, AST_CONTROL_UNHOLD);
+		if (!newchan) {
+			/* No party C or could not callback party B. */
+			return -1;
+		}
+
+		/* newchan is up, we should prepare transferee and bridge them */
+		if (ast_check_hangup(newchan)) {
 			ast_hangup(newchan);
 			return -1;
 		}
-		/* Make formats okay */
-		xferchan->visible_indication = transferer->visible_indication;
-		xferchan->readformat = transferee->readformat;
-		xferchan->writeformat = transferee->writeformat;
-		ast_channel_masquerade(xferchan, transferee);
-		ast_explicit_goto(xferchan, transferee->context, transferee->exten, transferee->priority);
-		xferchan->_state = AST_STATE_UP;
-		ast_clear_flag(xferchan, AST_FLAGS_ALL);
-		xferchan->_softhangup = 0;
-		if ((f = ast_read(xferchan)))
-			ast_frfree(f);
-		newchan->_state = AST_STATE_UP;
-		ast_clear_flag(newchan, AST_FLAGS_ALL);
-		newchan->_softhangup = 0;
-		if (!(tobj = ast_calloc(1, sizeof(*tobj)))) {
-			ast_hangup(xferchan);
-			ast_hangup(newchan);
+		if (check_compat(transferee, newchan)) {
 			return -1;
 		}
-
-		ast_channel_lock(newchan);
-		if ((features_datastore = ast_channel_datastore_find(newchan, &dial_features_info, NULL))) {
-				dialfeatures = features_datastore->data;
-		}
-		ast_channel_unlock(newchan);
-
-		if (dialfeatures) {
-			/* newchan should always be the callee and shows up as callee in dialfeatures, but for some reason
-			   I don't currently understand, the abilities of newchan seem to be stored on the caller side */
-			ast_copy_flags(&(config->features_callee), &(dialfeatures->features_caller), AST_FLAGS_ALL);
-			dialfeatures = NULL;
-		}
-
-		ast_channel_lock(xferchan);
-		if ((features_datastore = ast_channel_datastore_find(xferchan, &dial_features_info, NULL))) {
-			dialfeatures = features_datastore->data;
-		}
-		ast_channel_unlock(xferchan);
-	 
-		if (dialfeatures) {
-			ast_copy_flags(&(config->features_caller), &(dialfeatures->features_caller), AST_FLAGS_ALL);
-		}
-	 
-		tobj->chan = newchan;
-		tobj->peer = xferchan;
-		tobj->bconfig = *config;
-
-		if (tobj->bconfig.end_bridge_callback_data_fixup) {
-			tobj->bconfig.end_bridge_callback_data_fixup(&tobj->bconfig, tobj->peer, tobj->chan);
-		}
-
-		if (ast_stream_and_wait(newchan, xfersound, ""))
-			ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
-		bridge_call_thread_launch(tobj);
-		return -1;      /* XXX meaning the channel is bridged ? */
-	} else if (!ast_check_hangup(transferee)) {
-		/* act as blind transfer */
-		if (ast_autoservice_stop(transferee) < 0) {
-			ast_hangup(newchan);
-			return -1;
-		}
-
-		if (!newchan) {
-			unsigned int tries = 0;
-			char *transferer_tech, *transferer_name = ast_strdupa(transferer->name);
-
-			transferer_tech = strsep(&transferer_name, "/");
-			transferer_name = strsep(&transferer_name, "-");
-
-			if (ast_strlen_zero(transferer_name) || ast_strlen_zero(transferer_tech)) {
-				ast_log(LOG_WARNING, "Transferer has invalid channel name: '%s'\n", transferer->name);
-				if (ast_stream_and_wait(transferee, "beeperr", ""))
-					return -1;
-				return AST_FEATURE_RETURN_SUCCESS;
-			}
-
-			ast_log(LOG_NOTICE, "We're trying to call %s/%s\n", transferer_tech, transferer_name);
-			newchan = feature_request_and_dial(transferee, NULL, transferer_tech, ast_best_codec(transferee->nativeformats),
-				transferer_name, atxfernoanswertimeout, &outstate, transferee->cid.cid_num, transferee->cid.cid_name, 0, transferer->language);
-			while (!newchan && !atxferdropcall && tries < atxfercallbackretries) {
-				/* Trying to transfer again */
-				ast_autoservice_start(transferee);
-				ast_autoservice_ignore(transferee, AST_FRAME_DTMF_END);
-				ast_indicate(transferee, AST_CONTROL_HOLD);
-
-				newchan = feature_request_and_dial(transferer, transferee, "Local", ast_best_codec(transferer->nativeformats),
-					xferto, atxfernoanswertimeout, &outstate, transferer->cid.cid_num, transferer->cid.cid_name, 1, transferer->language);
-				if (ast_autoservice_stop(transferee) < 0) {
-					if (newchan)
-						ast_hangup(newchan);
-					return -1;
-				}
-				if (!newchan) {
-					/* Transfer failed, sleeping */
-					ast_debug(1, "Sleeping for %d ms before callback.\n", atxferloopdelay);
-					ast_safe_sleep(transferee, atxferloopdelay);
-					ast_debug(1, "Trying to callback...\n");
-					newchan = feature_request_and_dial(transferee, NULL, transferer_tech, ast_best_codec(transferee->nativeformats),
-						transferer_name, atxfernoanswertimeout, &outstate, transferee->cid.cid_num, transferee->cid.cid_name, 0, transferer->language);
-				}
-				tries++;
-			}
-		}
-		if (!newchan)
-			return -1;
-
-		/* newchan is up, we should prepare transferee and bridge them */
-		if (check_compat(transferee, newchan)) {
-			finishup(transferee);
-			return -1;
-		}
-		ast_indicate(transferee, AST_CONTROL_UNHOLD);
-
-		if ((ast_waitfordigit(transferee, 100) < 0)
-		   || (ast_waitfordigit(newchan, 100) < 0)
-		   || ast_check_hangup(transferee)
-		   || ast_check_hangup(newchan)) {
-			ast_hangup(newchan);
-			return -1;
-		}
-
-		xferchan = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, "Transfered/%s", transferee->name);
-		if (!xferchan) {
-			ast_hangup(newchan);
-			return -1;
-		}
-		/* Make formats okay */
-		xferchan->visible_indication = transferer->visible_indication;
-		xferchan->readformat = transferee->readformat;
-		xferchan->writeformat = transferee->writeformat;
-		ast_channel_masquerade(xferchan, transferee);
-		ast_explicit_goto(xferchan, transferee->context, transferee->exten, transferee->priority);
-		xferchan->_state = AST_STATE_UP;
-		ast_clear_flag(xferchan, AST_FLAGS_ALL);
-		xferchan->_softhangup = 0;
-		if ((f = ast_read(xferchan)))
-			ast_frfree(f);
-		newchan->_state = AST_STATE_UP;
-		ast_clear_flag(newchan, AST_FLAGS_ALL);
-		newchan->_softhangup = 0;
-		if (!(tobj = ast_calloc(1, sizeof(*tobj)))) {
-			ast_hangup(xferchan);
-			ast_hangup(newchan);
-			return -1;
-		}
-		tobj->chan = newchan;
-		tobj->peer = xferchan;
-		tobj->bconfig = *config;
-
-		if (tobj->bconfig.end_bridge_callback_data_fixup) {
-			tobj->bconfig.end_bridge_callback_data_fixup(&tobj->bconfig, tobj->peer, tobj->chan);
-		}
-
-		if (ast_stream_and_wait(newchan, xfersound, ""))
-			ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
-		bridge_call_thread_launch(tobj);
-		return -1;      /* XXX meaning the channel is bridged ? */
 	} else {
-		/* Transferee hung up */
-		finishup(transferee);
-		/* At this point both the transferer transferee have hungup,
-		 * so if newchan is up, hang it up as it has no one to talk to */
+		/*
+		 * Both the transferer and transferee have hungup.  If newchan
+		 * is up, hang it up as it has no one to talk to.
+		 */
+		ast_debug(1, "Everyone is hungup.\n");
 		if (newchan) {
 			ast_hangup(newchan);
 		}
 		return -1;
 	}
+
+	/* Initiate the channel transfer of party A to party C (or recalled party B). */
+
+	xferchan = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, "Transfered/%s", transferee->name);
+	if (!xferchan) {
+		ast_hangup(newchan);
+		return -1;
+	}
+
+	/* Give party A a momentary ringback tone during transfer. */
+	xferchan->visible_indication = AST_CONTROL_RINGING;
+
+	/* Make formats okay */
+	xferchan->readformat = transferee->readformat;
+	xferchan->writeformat = transferee->writeformat;
+
+	ast_channel_masquerade(xferchan, transferee);
+	ast_explicit_goto(xferchan, transferee->context, transferee->exten, transferee->priority);
+	xferchan->_state = AST_STATE_UP;
+	ast_clear_flag(xferchan, AST_FLAGS_ALL);
+
+	/* Do the masquerade manually to make sure that is is completed. */
+	ast_channel_lock(xferchan);
+	if (xferchan->masq) {
+		ast_do_masquerade(xferchan);
+	}
+	ast_channel_unlock(xferchan);
+
+	newchan->_state = AST_STATE_UP;
+	ast_clear_flag(newchan, AST_FLAGS_ALL);
+	tobj = ast_calloc(1, sizeof(*tobj));
+	if (!tobj) {
+		ast_hangup(xferchan);
+		ast_hangup(newchan);
+		return -1;
+	}
+
+	ast_channel_lock(newchan);
+	if ((features_datastore = ast_channel_datastore_find(newchan, &dial_features_info, NULL))) {
+		dialfeatures = features_datastore->data;
+	}
+	ast_channel_unlock(newchan);
+
+	if (dialfeatures) {
+		/* newchan should always be the callee and shows up as callee in dialfeatures, but for some reason
+		   I don't currently understand, the abilities of newchan seem to be stored on the caller side */
+		ast_copy_flags(&(config->features_callee), &(dialfeatures->features_caller), AST_FLAGS_ALL);
+		dialfeatures = NULL;
+	}
+
+	ast_channel_lock(xferchan);
+	if ((features_datastore = ast_channel_datastore_find(xferchan, &dial_features_info, NULL))) {
+		dialfeatures = features_datastore->data;
+	}
+	ast_channel_unlock(xferchan);
+
+	if (dialfeatures) {
+		ast_copy_flags(&(config->features_caller), &(dialfeatures->features_caller), AST_FLAGS_ALL);
+	}
+
+	tobj->chan = newchan;
+	tobj->peer = xferchan;
+	tobj->bconfig = *config;
+
+	if (tobj->bconfig.end_bridge_callback_data_fixup) {
+		tobj->bconfig.end_bridge_callback_data_fixup(&tobj->bconfig, tobj->peer, tobj->chan);
+	}
+
+	if (ast_stream_and_wait(newchan, xfersound, ""))
+		ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
+	bridge_call_thread_launch(tobj);
+	return -1;/* The transferee is masqueraded and the original bridged channels can be hungup. */
 }
 
 /* add atxfer and automon as undefined so you can only use em if you configure them */
@@ -2191,54 +2383,100 @@
 	}
 }
 
-/*! 
- * \brief Get feature and dial
- * \param caller,transferee,type,format,data,timeout,outstate,cid_num,cid_name,igncallerstate
+/*!
+ * \internal
+ * \brief Get feature and dial.
  *
- * Request channel, set channel variables, initiate call,check if they want to disconnect
- * go into loop, check if timeout has elapsed, check if person to be transfered hung up,
- * check for answer break loop, set cdr return channel.
+ * \param caller Channel to represent as the calling channel for the dialed channel.
+ * \param caller_name Original caller channel name.
+ * \param transferee Channel that the dialed channel will be transferred to.
+ * \param type Channel technology type to dial.
+ * \param format Codec formats for dialed channel.
+ * \param data Dialed channel extra parameters for ast_request() and ast_call().
+ * \param timeout Time limit for dialed channel to answer in ms. Must be greater than zero.
+ * \param outstate Status of dialed channel if unsuccessful.
+ * \param cid_num CallerID number to give dialed channel.
+ * \param cid_name CallerID name to give dialed channel.
+ * \param language Language of the caller.
  *
- * \todo XXX Check - this is very similar to the code in channel.c 
- * \return always a channel
-*/
-static struct ast_channel *feature_request_and_dial(struct ast_channel *caller, struct ast_channel *transferee, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, int igncallerstate, const char *language)
+ * \note
+ * outstate can be:
+ * 0, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION,
+ * AST_CONTROL_ANSWER, or AST_CONTROL_UNHOLD.  If
+ * AST_CONTROL_UNHOLD then the caller channel cancelled the
+ * transfer or the dialed channel did not answer before the
+ * timeout.
+ *
+ * \details
+ * Request channel, set channel variables, initiate call,
+ * check if they want to disconnect, go into loop, check if timeout has elapsed,
+ * check if person to be transfered hung up, check for answer break loop,
+ * set cdr return channel.
+ *
+ * \retval Channel Connected channel for transfer.
+ * \retval NULL on failure to get third party connected.
+ *
+ * \note This is similar to __ast_request_and_dial() in channel.c
+ */
+static struct ast_channel *feature_request_and_dial(struct ast_channel *caller,
+	const char *caller_name, struct ast_channel *transferee, const char *type,
+	int format, void *data, int timeout, int *outstate, const char *cid_num,
+	const char *cid_name, const char *language)
 {
 	int state = 0;
 	int cause = 0;
 	int to;
+	int caller_hungup;
+	int transferee_hungup;
 	struct ast_channel *chan;
-	struct ast_channel *monitor_chans[2];
+	struct ast_channel *monitor_chans[3];
 	struct ast_channel *active_channel;
-	int res = 0, ready = 0;
+	int ready = 0;
 	struct timeval started;
 	int x, len = 0;
 	char *disconnect_code = NULL, *dialed_code = NULL;
+	struct ast_frame *f;
+	AST_LIST_HEAD_NOLOCK(, ast_frame) deferred_frames;
+
+	caller_hungup = ast_check_hangup(caller);
 
 	if (!(chan = ast_request(type, format, data, &cause))) {
 		ast_log(LOG_NOTICE, "Unable to request channel %s/%s\n", type, (char *)data);
-		switch(cause) {
+		switch (cause) {
 		case AST_CAUSE_BUSY:
 			state = AST_CONTROL_BUSY;
 			break;
 		case AST_CAUSE_CONGESTION:
 			state = AST_CONTROL_CONGESTION;
 			break;
+		default:
+			state = 0;
+			break;
 		}
 		goto done;
 	}
 
 	ast_set_callerid(chan, cid_num, cid_name, cid_num);
 	ast_string_field_set(chan, language, language);
-	ast_channel_inherit_variables(caller, chan);	
-	pbx_builtin_setvar_helper(chan, "TRANSFERERNAME", caller->name);
-		
+	ast_channel_inherit_variables(caller, chan);
+	pbx_builtin_setvar_helper(chan, "TRANSFERERNAME", caller_name);
+
 	if (ast_call(chan, data, timeout)) {
 		ast_log(LOG_NOTICE, "Unable to call channel %s/%s\n", type, (char *)data);
+		switch (chan->hangupcause) {
+		case AST_CAUSE_BUSY:
+			state = AST_CONTROL_BUSY;
+			break;
+		case AST_CAUSE_CONGESTION:
+			state = AST_CONTROL_CONGESTION;
+			break;
+		default:
+			state = 0;
+			break;
+		}
 		goto done;
 	}
-	
-	ast_indicate(caller, AST_CONTROL_RINGING);
+
 	/* support dialing of the featuremap disconnect code while performing an attended tranfer */
 	ast_rwlock_rdlock(&features_lock);
 	for (x = 0; x < FEATURES_COUNT; x++) {
@@ -2255,57 +2493,111 @@
 	x = 0;
 	started = ast_tvnow();
 	to = timeout;
+	AST_LIST_HEAD_INIT_NOLOCK(&deferred_frames);
 
 	ast_poll_channel_add(caller, chan);
 
-	while (!((transferee && ast_check_hangup(transferee)) && (!igncallerstate && ast_check_hangup(caller))) && timeout && (chan->_state != AST_STATE_UP)) {
-		struct ast_frame *f = NULL;
-
-		monitor_chans[0] = caller;
-		monitor_chans[1] = chan;
-		active_channel = ast_waitfor_n(monitor_chans, 2, &to);
+	transferee_hungup = 0;
+	while (!ast_check_hangup(transferee) && (chan->_state != AST_STATE_UP)) {
+		int num_chans = 0;
+
+		monitor_chans[num_chans++] = transferee;
+		monitor_chans[num_chans++] = chan;
+		if (!caller_hungup) {
+			if (ast_check_hangup(caller)) {
+				caller_hungup = 1;
+
+#if defined(ATXFER_NULL_TECH)
+				/* Change caller's name to ensure that it will remain unique. */
+				set_new_chan_name(caller);
+
+				/*
+				 * Get rid of caller's physical technology so it is free for
+				 * other calls.
+				 */
+				set_null_chan_tech(caller);
+#endif	/* defined(ATXFER_NULL_TECH) */
+			} else {
+				/* caller is not hungup so monitor it. */
+				monitor_chans[num_chans++] = caller;
+			}
+		}
 
 		/* see if the timeout has been violated */
-		if(ast_tvdiff_ms(ast_tvnow(), started) > timeout) {
+		if (ast_tvdiff_ms(ast_tvnow(), started) > timeout) {
 			state = AST_CONTROL_UNHOLD;
-			ast_log(LOG_NOTICE, "We exceeded our AT-timeout\n");
+			ast_log(LOG_NOTICE, "We exceeded our AT-timeout for %s\n", chan->name);
 			break; /*doh! timeout*/
 		}
 
+		active_channel = ast_waitfor_n(monitor_chans, num_chans, &to);
 		if (!active_channel)
 			continue;
 
-		if (chan && (chan == active_channel)){
+		f = NULL;
+		if (transferee == active_channel) {
+			struct ast_frame *dup_f;
+
+			f = ast_read(transferee);
+			if (f == NULL) { /*doh! where'd he go?*/
+				transferee_hungup = 1;
+				state = 0;
+				break;
+			}
+			if (ast_is_deferrable_frame(f)) {
+				dup_f = ast_frisolate(f);
+				if (dup_f) {
+					if (dup_f == f) {
+						f = NULL;
+					}
+					AST_LIST_INSERT_HEAD(&deferred_frames, dup_f, frame_list);
+				}
+			}
+		} else if (chan == active_channel) {
 			if (!ast_strlen_zero(chan->call_forward)) {
-				if (!(chan = ast_call_forward(caller, chan, NULL, format, NULL, outstate))) {
-					return NULL;
+				state = 0;
+				chan = ast_call_forward(caller, chan, NULL, format, NULL, &state);
+				if (!chan) {
+					break;
 				}
 				continue;
 			}
 			f = ast_read(chan);
 			if (f == NULL) { /*doh! where'd he go?*/
-				state = AST_CONTROL_HANGUP;
-				res = 0;
+				switch (chan->hangupcause) {
+				case AST_CAUSE_BUSY:
+					state = AST_CONTROL_BUSY;
+					break;
+				case AST_CAUSE_CONGESTION:
+					state = AST_CONTROL_CONGESTION;
+					break;
+				default:
+					state = 0;
+					break;
+				}
 				break;
 			}
-			
-			if (f->frametype == AST_FRAME_CONTROL || f->frametype == AST_FRAME_DTMF || f->frametype == AST_FRAME_TEXT) {
+
+			if (f->frametype == AST_FRAME_CONTROL) {
 				if (f->subclass == AST_CONTROL_RINGING) {
-					state = f->subclass;
 					ast_verb(3, "%s is ringing\n", chan->name);
 					ast_indicate(caller, AST_CONTROL_RINGING);

[... 114 lines stripped ...]



More information about the asterisk-commits mailing list