[asterisk-commits] rmudgett: branch 1.6.2 r302173 - in /branches/1.6.2: ./ main/features.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 18 12:07:25 CST 2011
Author: rmudgett
Date: Tue Jan 18 12:07:15 2011
New Revision: 302173
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=302173
Log:
Merged revisions 302172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines
Issues with DTMF triggered attended transfers.
Issue #17999
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 #17096)
Problem: When A and B hangup, C is still ringing.
Issue #18395
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 #17273
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 #17273 and #18395 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 #17999)
Reported by: iskatel
Tested by: rmudgett
JIRA SWP-2246
(closes issue #17096)
Reported by: gelo
Tested by: rmudgett
JIRA SWP-1192
(closes issue #18395)
Reported by: shihchuan
Tested by: rmudgett
(closes issue #17273)
Reported by: grecco
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1047/
........
Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/main/features.c
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: branches/1.6.2/main/features.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/main/features.c?view=diff&rev=302173&r1=302172&r2=302173
==============================================================================
--- branches/1.6.2/main/features.c (original)
+++ branches/1.6.2/main/features.c Tue Jan 18 12:07:15 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);
- } else if ((f->subclass == AST_CONTROL_BUSY) || (f->subclass == AST_CONTROL_CONGESTION)) {
+ } else if (f->subclass == AST_CONTROL_BUSY) {
state = f->subclass;
ast_verb(3, "%s is busy\n", chan->name);
ast_indicate(caller, AST_CONTROL_BUSY);
ast_frfree(f);
- f = NULL;
+ break;
+ } else if (f->subclass == AST_CONTROL_CONGESTION) {
+ state = f->subclass;
+ ast_verb(3, "%s is congested\n", chan->name);
+ ast_indicate(caller, AST_CONTROL_CONGESTION);
+ ast_frfree(f);
break;
} else if (f->subclass == AST_CONTROL_ANSWER) {
/* This is what we are hoping for */
state = f->subclass;
ast_frfree(f);
- f = NULL;
ready=1;
break;
} else if (f->subclass != -1 && f->subclass != AST_CONTROL_PROGRESS) {
@@ -2315,22 +2607,9 @@
} else if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_VIDEO) {
ast_write(caller, f);
}
-
- } else if (caller && (active_channel == caller)) {
+ } else if (caller == active_channel) {
f = ast_read(caller);
- if (f == NULL) { /*doh! where'd he go?*/
- if (!igncallerstate) {
- if (ast_check_hangup(caller) && !ast_check_hangup(chan)) {
- /* make this a blind transfer */
- ready = 1;
- break;
- }
- state = AST_CONTROL_HANGUP;
- res = 0;
- break;
- }
- } else {
-
+ if (f) {
if (f->frametype == AST_FRAME_DTMF) {
dialed_code[x++] = f->subclass;
dialed_code[x] = '\0';
@@ -2344,7 +2623,6 @@
/* Caller Canceled the call */
state = AST_CONTROL_UNHOLD;
ast_frfree(f);
- f = NULL;
break;
}
} else if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_VIDEO) {
@@ -2357,21 +2635,31 @@
} /* end while */
ast_poll_channel_del(caller, chan);
-
+
+ /*
+ * We need to free all the deferred frames, but we only need to
+ * queue the deferred frames if no hangup was received.
+ */
+ ast_channel_lock(transferee);
+ transferee_hungup = (transferee_hungup || ast_check_hangup(transferee));
+ while ((f = AST_LIST_REMOVE_HEAD(&deferred_frames, frame_list))) {
[... 28 lines stripped ...]
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