[svn-commits] branch north/bug_1082 - r8222 in
/team/north/bug_1082: apps/ channels/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed Jan 18 16:45:47 MST 2006
Author: north
Date: Wed Jan 18 17:45:46 2006
New Revision: 8222
URL: http://svn.digium.com/view/asterisk?rev=8222&view=rev
Log:
Took patch from bug 1082, changed it to set channel var from queues, and read same var from agent_call
Modified:
team/north/bug_1082/apps/app_queue.c
team/north/bug_1082/channels/chan_agent.c
Modified: team/north/bug_1082/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/north/bug_1082/apps/app_queue.c?rev=8222&r1=8221&r2=8222&view=diff
==============================================================================
--- team/north/bug_1082/apps/app_queue.c (original)
+++ team/north/bug_1082/apps/app_queue.c Wed Jan 18 17:45:46 2006
@@ -350,6 +350,7 @@
char sound_thanks[80]; /*!< Sound file: "Thank you for your patience." (def. queue-thankyou) */
char sound_reporthold[80]; /*!< Sound file: "Hold time" (def. queue-reporthold) */
char sound_periodicannounce[MAX_PERIODIC_ANNOUNCEMENTS][80];/* Sound files: Custom announce, no default */
+ char sound_ack[80]; /*!< Sound file: Per queue agent ack sound, no default */
int count; /*!< How many entries */
int maxlen; /*!< Max number of entries */
@@ -689,6 +690,8 @@
}
} else if (!strcasecmp(param, "periodic-announce-frequency")) {
q->periodicannouncefrequency = atoi(val);
+ } else if (!strcasecmp(param, "acksound")) {
+ ast_copy_string(q->sound_ack, val, sizeof(q->sound_ack));
} else if (!strcasecmp(param, "retry")) {
q->retry = atoi(val);
if (q->retry < 0)
@@ -1433,6 +1436,9 @@
/* Presense of ADSI CPE on outgoing channel follows ours */
tmp->chan->adsicpe = qe->chan->adsicpe;
+
+ /* Allows agent to play a per queue ack sound */
+ pbx_builtin_setvar_helper(tmp->chan, "QUEUEACKSOUND", qe->parent->sound_ack);
/* Place the call, but don't wait on the answer */
res = ast_call(tmp->chan, location, 0);
Modified: team/north/bug_1082/channels/chan_agent.c
URL: http://svn.digium.com/view/asterisk/team/north/bug_1082/channels/chan_agent.c?rev=8222&r1=8221&r2=8222&view=diff
==============================================================================
--- team/north/bug_1082/channels/chan_agent.c (original)
+++ team/north/bug_1082/channels/chan_agent.c Wed Jan 18 17:45:46 2006
@@ -54,6 +54,7 @@
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
+#include "asterisk/translate.h"
#include "asterisk/lock.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
@@ -157,6 +158,8 @@
static int autologoff;
static int wrapuptime;
static int ackcall;
+static char acksound[AST_MAX_BUF] = "";
+static int ackwait = 0;
static int multiplelogin = 1;
static int autologoffunavail = 0;
@@ -206,6 +209,8 @@
char loginchan[80]; /**< channel they logged in from */
char logincallerid[80]; /**< Caller ID they had when they logged in */
struct ast_channel *chan; /**< Channel we use */
+ struct ast_trans_pvt *trans;
+ int framecount;
struct agent_pvt *next; /**< Next Agent in the linked list. */
};
@@ -501,6 +506,14 @@
/* Note that we don't hangup if it's not a callback because Asterisk will do it
for us when the PBX instance that called login finishes */
if (!ast_strlen_zero(p->loginchan)) {
+ if(p && p->chan && p->chan->stream) {
+ ast_closestream(p->chan->stream);
+ p->chan->stream = NULL;
+ }
+ if(p && p->trans) {
+ ast_translator_free_path(p->trans);
+ p->trans = NULL;
+ }
if (p->chan)
ast_log(LOG_DEBUG, "Bridge on '%s' being cleared (2)\n", p->chan->name);
@@ -529,6 +542,7 @@
if (p->ackcall) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s answered, waiting for '#' to acknowledge\n", p->chan->name);
+ p->framecount = 1;
/* Don't pass answer along */
ast_frfree(f);
f = &null_frame;
@@ -543,6 +557,14 @@
break;
case AST_FRAME_DTMF:
if (!p->acknowledged && (f->subclass == '#')) {
+ if(p && p->chan && p->chan->stream) {
+ ast_closestream(p->chan->stream);
+ p->chan->stream = NULL;
+ }
+ if(p && p->trans) {
+ ast_translator_free_path(p->trans);
+ p->trans = NULL;
+ }
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s acknowledged\n", p->chan->name);
p->acknowledged = 1;
@@ -664,6 +686,7 @@
static int agent_call(struct ast_channel *ast, char *dest, int timeout)
{
struct agent_pvt *p = ast->tech_pvt;
+ struct ast_frame *f2 = NULL;
int res = -1;
int newstate=0;
ast_mutex_lock(&p->lock);
@@ -735,6 +758,57 @@
}
res = 0;
}
+
+ /* Turn this code into poor man's generator since we are reading frames at the perfect speed
+ just open a stream and pass along the frames from the file while we are waiting for
+ the agent to press # or the timeout to occur.
+ */
+ if(p && ! p->acknowledged) {
+ /* We don't want to rely on queues being loaded,
+ so we'll just use a channel variable that gets set from queues.
+ */
+ char *oldacksound = NULL;
+ if (!ast_strlen_zero(acksound))
+ oldacksound = ast_strdupa(acksound);
+ if (!ast_strlen_zero(pbx_builtin_getvar_helper(ast, "QUEUEACKSOUND")))
+ strncpy(acksound, pbx_builtin_getvar_helper(ast, "QUEUEACKSOUND"), AST_MAX_BUF);
+
+ if(p->chan && !ast_strlen_zero(acksound) && p->framecount++ > ackwait &&
+ ! p->trans && (ast_openstream(p->chan,acksound, p->chan->language))) {
+ if((f2 = ast_readframe(p->chan->stream))) {
+ if(p->chan->readformat && p->chan->readformat != f2->subclass)
+ p->trans = ast_translator_build_path(p->chan->readformat, f2->subclass);
+ }
+ }
+ ast_copy_string(acksound, oldacksound, sizeof(acksound));
+
+ if(p->framecount > ackwait && p->chan && p->chan->stream) {
+ if(!f2) {
+ if(!(f2 = ast_readframe(p->chan->stream))) {
+ /* Reset the stream and the ackwait and play the file again if the time comes. */
+ if(p->chan && p->chan->stream) {
+ ast_closestream(p->chan->stream);
+ p->chan->stream = NULL;
+ }
+ if(p->trans) {
+ ast_translator_free_path(p->trans);
+ p->trans = NULL;
+ }
+ p->framecount = 0;
+ }
+ }
+ if(f2) {
+ if(p->trans && f2->subclass != p->chan->readformat)
+ f2 = ast_translate(p->trans, f2, 1);
+ if(f2) {
+ ast_write(p->chan, f2);
+ ast_frfree(f2);
+ f2 = NULL;
+ }
+ }
+ }
+ }
+
CLEANUP(ast,p);
ast_mutex_unlock(&p->lock);
if (newstate)
@@ -765,6 +839,15 @@
ast->tech_pvt = NULL;
p->app_sleep_cond = 1;
p->acknowledged = 0;
+
+ if(p && p->chan && p->chan->stream) {
+ ast_stopstream(p->chan);
+ p->chan->stream = NULL;
+ }
+ if(p && p->trans) {
+ ast_translator_free_path(p->trans);
+ p->trans = NULL;
+ }
/* if they really are hung up then set start to 0 so the test
* later if we're called on an already downed channel
@@ -991,6 +1074,8 @@
* can safely use signals for this purpose. The pselect() needs to be
* implemented in the kernel for this.
*/
+ p->trans = NULL;
+ p->framecount = 0;
p->app_sleep_cond = 0;
if( ast_mutex_trylock(&p->app_lock) )
{
@@ -1076,6 +1161,10 @@
/* Create the interface list */
if (!strcasecmp(v->name, "agent")) {
add_agent(v->value, 0);
+ } else if (!strcasecmp(v->name, "acksound")) {
+ strncpy(acksound,v->value,AST_MAX_BUF);
+ } else if (!strcasecmp(v->name, "ackwait")) {
+ ackwait = atoi(v->value);
} else if (!strcasecmp(v->name, "group")) {
group = ast_get_group(v->value);
} else if (!strcasecmp(v->name, "autologoff")) {
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