Version 0.3.0 from FTP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@556 f38db490-d61c-443f-a65b-d21fe96a405b

15 files changed, 124 insertions, 86 deletions

diff --git a/apps/app_disa.c b/apps/app_disa.c
index a105c9ed8..89d956deb 100755
--- a/apps/app_disa.c
+++ b/apps/app_disa.c
@@ -116,12 +116,13 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	int i,j,k,x;
 	struct localuser *u;
 	char tmp[256],arg2[256],exten[AST_MAX_EXTENSION],acctcode[20];
-	unsigned char tone_block[640],sil_block[640];
+	unsigned char tone_block[640];
 	char *ourcontext,*ourcallerid;
 	struct ast_frame *f,wf;
 	struct timeval lastout, now, lastdigittime;
 	int res;
 	FILE *fp;
+	char *stringp=NULL;
 
 	if (ast_set_write_format(chan,AST_FORMAT_ULAW))
 	{
@@ -134,19 +135,18 @@ static int disa_exec(struct ast_channel *chan, void *data)
 		return -1;
 	}
 	lastout.tv_sec = lastout.tv_usec = 0;
-	  /* make block of silence */
-	memset(sil_block,0x7f,sizeof(sil_block));
 	if (!data || !strlen((char *)data)) {
 		ast_log(LOG_WARNING, "disa requires an argument (passcode/passcode file)\n");
 		return -1;
 	}
 	strncpy(tmp, (char *)data, sizeof(tmp)-1);
-	strtok(tmp, "|");
-	ourcontext = strtok(NULL, "|");
+	stringp=tmp;
+	strsep(&stringp, "|");
+	ourcontext = strsep(&stringp, "|");
 	/* if context specified, save 2nd arg and parse third */
 	if (ourcontext) {
 		strcpy(arg2,ourcontext);
-		ourcallerid = strtok(NULL,"|");
+		ourcallerid = strsep(&stringp,"|");
 	}
 	  /* if context not specified, use "disa" */
 	else {
@@ -162,6 +162,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 	}
 	i = k = x = 0; /* k is 0 for pswd entry, 1 for ext entry */
 	exten[0] = 0;
+	acctcode[0] = 0;
 	/* can we access DISA without password? */ 
 	if (!strcasecmp(tmp, "no-password"))
 	{
@@ -207,7 +208,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 				wf.data = tone_block;
 				wf.datalen = f->datalen;
 				make_tone_block(tone_block, 350, 440, f->datalen, &x);
-				wf.timelen = wf.datalen / 8;
+				wf.samples = wf.datalen;
 				ast_frfree(f);
 			    if (ast_write(chan, &wf)) 
 				{
@@ -250,6 +251,7 @@ static int disa_exec(struct ast_channel *chan, void *data)
 						tmp[0] = 0;
 						while(fgets(tmp,sizeof(tmp) - 1,fp))
 						   {
+							char *stringp=NULL;
 							if (!tmp[0]) continue;
 							if (tmp[strlen(tmp) - 1] == '\n') 
 								tmp[strlen(tmp) - 1] = 0;
@@ -257,10 +259,11 @@ static int disa_exec(struct ast_channel *chan, void *data)
 							  /* skip comments */
 							if (tmp[0] == '#') continue;
 							if (tmp[0] == ';') continue;
-							strtok(tmp, "|");
+							stringp=tmp;
+							strsep(&stringp, "|");
 							/* save 2nd arg as clid */
 							ourcallerid = arg2;
-							ourcontext = strtok(NULL, "|");
+							ourcontext = strsep(&stringp, "|");
 							  /* password must be in valid format (numeric) */
 							if (sscanf(tmp,"%d",&j) < 1) continue;
 							  /* if we got it */
@@ -328,7 +331,7 @@ reorder:
 				wf.mallocd = 0;
 				wf.data = tone_block;
 				wf.datalen = f->datalen;
-				wf.timelen = wf.datalen / 8;
+				wf.samples = wf.datalen;
 				if (k) 
 					memset(tone_block, 0x7f, wf.datalen);
 				else
diff --git a/apps/app_festival.c b/apps/app_festival.c
index d45988c99..1e88fff8e 100755
--- a/apps/app_festival.c
+++ b/apps/app_festival.c
@@ -211,7 +211,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
 					myf.f.frametype = AST_FRAME_VOICE;
 					myf.f.subclass = AST_FORMAT_SLINEAR;
 					myf.f.datalen = res;
-					myf.f.timelen = res / 16;
+					myf.f.samples = res / 2;
 					myf.f.mallocd = 0;
 					myf.f.offset = AST_FRIENDLY_OFFSET;
 					myf.f.src = __PRETTY_FUNCTION__;
diff --git a/apps/app_mp3.c b/apps/app_mp3.c
index 5ef5f9bb0..e99415d53 100755
--- a/apps/app_mp3.c
+++ b/apps/app_mp3.c
@@ -164,7 +164,7 @@ static int mp3_exec(struct ast_channel *chan, void *data)
 					myf.f.frametype = AST_FRAME_VOICE;
 					myf.f.subclass = AST_FORMAT_SLINEAR;
 					myf.f.datalen = res;
-					myf.f.timelen = res / 16;
+					myf.f.samples = res / 2;
 					myf.f.mallocd = 0;
 					myf.f.offset = AST_FRIENDLY_OFFSET;
 					myf.f.src = __PRETTY_FUNCTION__;
diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index 3e35a821f..225c8ba74 100755
--- a/channels/chan_alsa.c
+++ b/channels/chan_alsa.c
@@ -646,7 +646,7 @@ static struct ast_frame *alsa_read(struct ast_channel *chan)
 	
 	f.frametype = AST_FRAME_NULL;
 	f.subclass = 0;
-	f.timelen = 0;
+	f.samples = 0;
 	f.datalen = 0;
 	f.data = NULL;
 	f.offset = 0;
@@ -728,7 +728,7 @@ static struct ast_frame *alsa_read(struct ast_channel *chan)
 		}
 		f.frametype = AST_FRAME_VOICE;
 		f.subclass = AST_FORMAT_SLINEAR;
-		f.timelen = FRAME_SIZE / 8;
+		f.samples = FRAME_SIZE;
 		f.datalen = FRAME_SIZE * 2;
 		f.data = buf;
 		f.offset = AST_FRIENDLY_OFFSET;
@@ -905,8 +905,8 @@ static char sendtext_usage[] =
 
 static int console_sendtext(int fd, int argc, char *argv[])
 {
-	int tmparg = 1;
-	if (argc < 1)
+	int tmparg = 2;
+	if (argc < 2)
 		return RESULT_SHOWUSAGE;
 	if (!alsa.owner) {
 		ast_cli(fd, "No one is calling us\n");
@@ -968,9 +968,11 @@ static int console_dial(int fd, int argc, char *argv[])
 	mye = exten;
 	myc = context;
 	if (argc == 2) {
+		char *stringp=NULL;
 		strncpy(tmp, argv[1], sizeof(tmp)-1);
-		strtok(tmp, "@");
-		tmp2 = strtok(NULL, "@");
+		stringp=tmp;
+		strsep(&stringp, "@");
+		tmp2 = strsep(&stringp, "@");
 		if (strlen(tmp))
 			mye = tmp;
 		if (tmp2 && strlen(tmp2))
@@ -995,7 +997,7 @@ static struct ast_cli_entry myclis[] = {
 	{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
 	{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
 	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
-	{ { "send text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
+	{ { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
 	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
 };
 
diff --git a/channels/chan_modem_aopen.c b/channels/chan_modem_aopen.c
index b33ce074c..a838671de 100755
--- a/channels/chan_modem_aopen.c
+++ b/channels/chan_modem_aopen.c
@@ -187,7 +187,7 @@ static struct ast_frame *aopen_handle_escape(struct ast_modem_pvt *p, char esc)
 	p->fr.subclass = 0;
 	p->fr.data = NULL;
 	p->fr.datalen = 0;
-	p->fr.timelen = 0;
+	p->fr.samples = 0;
 	p->fr.offset = 0;
 	p->fr.mallocd = 0;
 	if (esc)
@@ -315,7 +315,7 @@ static struct ast_frame *aopen_read(struct ast_modem_pvt *p)
 		/* If we get here, we have a complete voice frame */
 		p->fr.frametype = AST_FRAME_VOICE;
 		p->fr.subclass = AST_FORMAT_SLINEAR;
-		p->fr.timelen = 30;
+		p->fr.samples = 240;
 		p->fr.data = p->obuf;
 		p->fr.datalen = p->obuflen;
 		p->fr.mallocd = 0;
diff --git a/channels/chan_modem_bestdata.c b/channels/chan_modem_bestdata.c
index 08a1c2e89..7ec76aa5e 100755
--- a/channels/chan_modem_bestdata.c
+++ b/channels/chan_modem_bestdata.c
@@ -160,7 +160,7 @@ static struct ast_frame *bestdata_handle_escape(struct ast_modem_pvt *p, char es
 	p->fr.subclass = 0;
 	p->fr.data = NULL;
 	p->fr.datalen = 0;
-	p->fr.timelen = 0;
+	p->fr.samples = 0;
 	p->fr.offset = 0;
 	p->fr.mallocd = 0;
 	if (esc)
@@ -364,7 +364,7 @@ static struct ast_frame *bestdata_read(struct ast_modem_pvt *p)
 		/* If we get here, we have a complete voice frame */
 		p->fr.frametype = AST_FRAME_VOICE;
 		p->fr.subclass = AST_FORMAT_SLINEAR;
-		p->fr.timelen = 30;
+		p->fr.samples = 240;
 		p->fr.data = p->obuf;
 		p->fr.datalen = p->obuflen;
 		p->fr.mallocd = 0;
diff --git a/channels/chan_modem_i4l.c b/channels/chan_modem_i4l.c
index 553f144ee..18b5a5039 100755
--- a/channels/chan_modem_i4l.c
+++ b/channels/chan_modem_i4l.c
@@ -17,6 +17,7 @@
 #include <stdlib.h>
 #include <errno.h>
 #include <unistd.h>
+#include <sys/ioctl.h>
 #include <asterisk/lock.h>
 #include <asterisk/vmodem.h>
 #include <asterisk/module.h>
@@ -151,11 +152,37 @@ static int i4l_init(struct ast_modem_pvt *p)
 			return -1;
 		}
 	}
+	if (strlen(p->incomingmsn)) {
+		char *q;
+		snprintf(cmd, sizeof(cmd), "AT&L%s", p->incomingmsn);
+		// translate , into ; since that is the seperator I4L uses, but can't be directly
+		// put in the config file because it will interpret the rest of the line as comment.
+		q = cmd+4;
+		while (*q) {
+			if (*q == ',') *q = ';';
+			++q;
+		}
+		if (ast_modem_send(p, cmd, 0) ||
+		    ast_modem_expect(p, "OK", 5)) {
+			ast_log(LOG_WARNING, "Unable to set Listen to %s\n", p->msn);
+			return -1;
+		}
+	}
+	if (ast_modem_send(p, "AT&D2", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to DTR disconnect mode\n");
+		return -1;
+	}
 	if (ast_modem_send(p, "ATS18=1", 0) ||
 	     ast_modem_expect(p, "OK", 5)) {
 		ast_log(LOG_WARNING, "Unable to set to audio only mode\n");
 		return -1;
 	}
+	if (ast_modem_send(p, "ATS13.6=1", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to RUNG indication\n");
+		return -1;
+	}
 	if (ast_modem_send(p, "ATS14=4", 0) ||
 	     ast_modem_expect(p, "OK", 5)) {
 		ast_log(LOG_WARNING, "Unable to set to transparent mode\n");
@@ -189,7 +216,7 @@ static struct ast_frame *i4l_handle_escape(struct ast_modem_pvt *p, char esc)
 	p->fr.subclass = 0;
 	p->fr.data = NULL;
 	p->fr.datalen = 0;
-	p->fr.timelen = 0;
+	p->fr.samples = 0;
 	p->fr.offset = 0;
 	p->fr.mallocd = 0;
 	if (esc && option_debug)
@@ -271,7 +298,15 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
 	int x;
 	if (p->ministate == STATE_COMMAND) {
 		/* Read the first two bytes, first, in case it's a control message */
-		read(p->fd, result, 2);
+		res = read(p->fd, result, 2);
+		if (res < 2) {
+			// short read, means there was a hangup?
+			// (or is this also possible without hangup?)
+			// Anyway, reading from unitialized buffers is a bad idea anytime.
+			if (errno == EAGAIN)
+				return i4l_handle_escape(p, 0);
+			return NULL;
+		}
 		if (result[0] == CHAR_DLE) {
 			return i4l_handle_escape(p, result[1]);
 			
@@ -283,7 +318,7 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
 			ast_modem_trim(result);
 			if (!strcasecmp(result, "VCON")) {
 				/* If we're in immediate mode, reply now */
-				if (p->mode == MODEM_MODE_IMMEDIATE)
+//				if (p->mode == MODEM_MODE_IMMEDIATE)
 					return i4l_handle_escape(p, 'X');
 			} else
 			if (!strcasecmp(result, "BUSY")) {
@@ -292,16 +327,22 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
 			} else
 			if (!strncasecmp(result, "CALLER NUMBER: ", 15 )) {
 				strncpy(p->cid, result + 15, sizeof(p->cid)-1);
-				return i4l_handle_escape(p, 'R');
+				return i4l_handle_escape(p, 0);
 			} else
 			if (!strcasecmp(result, "RINGING")) {
 				if (option_verbose > 2)
 					ast_verbose(VERBOSE_PREFIX_3 "%s is ringing...\n", p->dev);
 				return i4l_handle_escape(p, 'I');
 			} else
+			if (!strncasecmp(result, "RUNG", 4)) {
+				/* PM2002: the line was hung up before we picked it up, bye bye */
+				if (option_verbose > 2) 
+					ast_verbose(VERBOSE_PREFIX_3 "%s was hung up on before we answered\n", p->dev);
+				return NULL;
+			} else
 			if (!strncasecmp(result, "RING", 4)) {
 				if (result[4]=='/') 
-					strncpy(p->dnid, result + 4, sizeof(p->dnid)-1);
+					strncpy(p->dnid, result + 5, sizeof(p->dnid)-1);
 				return i4l_handle_escape(p, 'R');
 			} else
 			if (!strcasecmp(result, "NO CARRIER")) {
@@ -329,6 +370,7 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
 				if (errno == EAGAIN)
 					return i4l_handle_escape(p, 0);
 				ast_log(LOG_WARNING, "Read failed: %s\n", strerror(errno));
+				return NULL;
 			}
 			
 			for (x=0;x<res;x++) {
@@ -372,7 +414,7 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
 		/* If we get here, we have a complete voice frame */
 		p->fr.frametype = AST_FRAME_VOICE;
 		p->fr.subclass = AST_FORMAT_SLINEAR;
-		p->fr.timelen = 30;
+		p->fr.samples = 240;
 		p->fr.data = p->obuf;
 		p->fr.datalen = p->obuflen;
 		p->fr.mallocd = 0;
@@ -497,32 +539,24 @@ static int i4l_dial(struct ast_modem_pvt *p, char *stuff)
 static int i4l_hangup(struct ast_modem_pvt *p)
 {
 	char dummy[50];
-	sprintf(dummy, "%c%c", 0x10, 0x3);
-	if (write(p->fd, dummy, 2) < 0) {
-		ast_log(LOG_WARNING, "Failed to break\n");
-		return -1;
-	}
+	int dtr = TIOCM_DTR;
+
+	/* down DTR to hangup modem */
+	ioctl(p->fd, TIOCMBIC, &dtr);
 
 	/* Read anything outstanding */
 	while(read(p->fd, dummy, sizeof(dummy)) > 0);
 
-	sprintf(dummy, "%c%c", 0x10, 0x14);
-	if (write(p->fd, dummy, 2) < 0) {
-		ast_log(LOG_WARNING, "Failed to break\n");
-		return -1;
-	}
-	ast_modem_expect(p, "VCON", 1);
-#if 0
-	if (ast_modem_expect(p, "VCON", 8)) {
-		ast_log(LOG_WARNING, "Didn't get expected VCON\n");
-		return -1;
-	}
-#endif
+	/* rise DTR to re-enable line */
+	ioctl(p->fd, TIOCMBIS, &dtr);
+	
+	/* Read anything outstanding */
+	while(read(p->fd, dummy, sizeof(dummy)) > 0);
+
+	/* basically we're done, just to be sure */
 	write(p->fd, "\n\n", 2);
 	read(p->fd, dummy, sizeof(dummy));
-	/* Hangup by switching to data, then back to voice */
-	if (ast_modem_send(p, "ATH", 0) ||
-	     ast_modem_expect(p, "NO CARRIER", 8)) {
+	if (ast_modem_send(p, "ATH", 0)) {
 		ast_log(LOG_WARNING, "Unable to hang up\n");
 		return -1;
 	}
@@ -530,6 +564,7 @@ static int i4l_hangup(struct ast_modem_pvt *p)
 		ast_log(LOG_WARNING, "Final 'OK' not received\n");
 		return -1;
 	}
+
 	return 0;
 }
 
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index 09f5b1b04..9a6bc9bcd 100755
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -320,7 +320,7 @@ static struct ast_frame  *phone_exception(struct ast_channel *ast)
 
 	/* Some nice norms */
 	p->fr.datalen = 0;
-	p->fr.timelen = 0;
+	p->fr.samples = 0;
 	p->fr.data =  NULL;
 	p->fr.src = type;
 	p->fr.offset = 0;
@@ -381,7 +381,7 @@ static struct ast_frame  *phone_read(struct ast_channel *ast)
 
 	/* Some nice norms */
 	p->fr.datalen = 0;
-	p->fr.timelen = 0;
+	p->fr.samples = 0;
 	p->fr.data =  NULL;
 	p->fr.src = type;
 	p->fr.offset = 0;
@@ -415,6 +415,7 @@ static struct ast_frame  *phone_read(struct ast_channel *ast)
 		res = 4;
 		break;
 	}
+	p->fr.samples = 240;
 	p->fr.datalen = res;
 	p->fr.frametype = AST_FRAME_VOICE;
 	p->fr.subclass = p->lastinput;
@@ -630,6 +631,8 @@ static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *conte
 		strncpy(tmp->context, context, sizeof(tmp->context)-1);
 		if (strlen(i->ext))
 			strncpy(tmp->exten, i->ext, sizeof(tmp->exten)-1);
+		else
+			strncpy(tmp->exten, "s",  sizeof(tmp->exten) - 1);
 		if (strlen(i->language))
 			strncpy(tmp->language, i->language, sizeof(tmp->language)-1);
 		if (strlen(i->callerid))
diff --git a/codecs/codec_a_mu.c b/codecs/codec_a_mu.c
index c7d52b447..d7d6ff11e 100755
--- a/codecs/codec_a_mu.c
+++ b/codecs/codec_a_mu.c
@@ -125,7 +125,7 @@ alawtoulaw_frameout (struct ast_translator_pvt *pvt)
   tmp->f.frametype = AST_FRAME_VOICE;
   tmp->f.subclass = AST_FORMAT_ULAW;
   tmp->f.datalen = tmp->tail;
-  tmp->f.timelen = tmp->tail / 8;
+  tmp->f.samples = tmp->tail;
   tmp->f.mallocd = 0;
   tmp->f.offset = AST_FRIENDLY_OFFSET;
   tmp->f.src = __PRETTY_FUNCTION__;
@@ -172,7 +172,7 @@ ulawtoalaw_frameout (struct ast_translator_pvt *pvt)
   if (tmp->tail) {
 	  tmp->f.frametype = AST_FRAME_VOICE;
 	  tmp->f.subclass = AST_FORMAT_ALAW;
-	  tmp->f.timelen = tmp->tail / 8;
+	  tmp->f.samples = tmp->tail;
 	  tmp->f.mallocd = 0;
 	  tmp->f.offset = AST_FRIENDLY_OFFSET;
 	  tmp->f.src = __PRETTY_FUNCTION__;
@@ -195,7 +195,7 @@ alawtoulaw_sample ()
   f.frametype = AST_FRAME_VOICE;
   f.subclass = AST_FORMAT_ALAW;
   f.datalen = sizeof (ulaw_slin_ex);
-  f.timelen = sizeof(ulaw_slin_ex) / 8;
+  f.samples = sizeof(ulaw_slin_ex);
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
@@ -210,7 +210,7 @@ ulawtoalaw_sample ()
   f.frametype = AST_FRAME_VOICE;
   f.subclass = AST_FORMAT_ULAW;
   f.datalen = sizeof (ulaw_slin_ex);
-  f.timelen = sizeof(ulaw_slin_ex) / 8;
+  f.samples = sizeof(ulaw_slin_ex);
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
diff --git a/codecs/codec_adpcm.c b/codecs/codec_adpcm.c
index c07aba538..fdd03675e 100755
--- a/codecs/codec_adpcm.c
+++ b/codecs/codec_adpcm.c
@@ -290,7 +290,7 @@ adpcmtolin_framein (struct ast_translator_pvt *pvt, struct ast_frame *f)
  *
  * Results:
  *  Converted signals are placed in tmp->f.data, tmp->f.datalen
- *  and tmp->f.timelen are calculated.
+ *  and tmp->f.samples are calculated.
  *
  * Side effects:
  *  None.
@@ -307,7 +307,7 @@ adpcmtolin_frameout (struct ast_translator_pvt *pvt)
   tmp->f.frametype = AST_FRAME_VOICE;
   tmp->f.subclass = AST_FORMAT_SLINEAR;
   tmp->f.datalen = tmp->tail *2;
-  tmp->f.timelen = tmp->tail / 8;
+  tmp->f.samples = tmp->tail;
   tmp->f.mallocd = 0;
   tmp->f.offset = AST_FRIENDLY_OFFSET;
   tmp->f.src = __PRETTY_FUNCTION__;
@@ -383,7 +383,7 @@ lintoadpcm_frameout (struct ast_translator_pvt *pvt)
 
   tmp->f.frametype = AST_FRAME_VOICE;
   tmp->f.subclass = AST_FORMAT_ADPCM;
-  tmp->f.timelen = i_max / 8;
+  tmp->f.samples = i_max;
   tmp->f.mallocd = 0;
   tmp->f.offset = AST_FRIENDLY_OFFSET;
   tmp->f.src = __PRETTY_FUNCTION__;
@@ -417,7 +417,7 @@ adpcmtolin_sample ()
   f.frametype = AST_FRAME_VOICE;
   f.subclass = AST_FORMAT_ADPCM;
   f.datalen = sizeof (adpcm_slin_ex);
-  f.timelen = sizeof(adpcm_slin_ex) / 4;
+  f.samples = sizeof(adpcm_slin_ex) * 2;
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
@@ -437,7 +437,7 @@ lintoadpcm_sample ()
   f.subclass = AST_FORMAT_SLINEAR;
   f.datalen = sizeof (slin_adpcm_ex);
   /* Assume 8000 Hz */
-  f.timelen = sizeof (slin_adpcm_ex) / 16;
+  f.samples = sizeof (slin_adpcm_ex) / 2;
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
diff --git a/codecs/codec_alaw.c b/codecs/codec_alaw.c
index 833092365..54a464cfc 100755
--- a/codecs/codec_alaw.c
+++ b/codecs/codec_alaw.c
@@ -151,7 +151,7 @@ alawtolin_framein (struct ast_translator_pvt *pvt, struct ast_frame *f)
  *
  * Results:
  *  Converted signals are placed in tmp->f.data, tmp->f.datalen
- *  and tmp->f.timelen are calculated.
+ *  and tmp->f.samples are calculated.
  *
  * Side effects:
  *  None.
@@ -168,7 +168,7 @@ alawtolin_frameout (struct ast_translator_pvt *pvt)
   tmp->f.frametype = AST_FRAME_VOICE;
   tmp->f.subclass = AST_FORMAT_SLINEAR;
   tmp->f.datalen = tmp->tail *2;
-  tmp->f.timelen = tmp->tail / 8;
+  tmp->f.samples = tmp->tail;
   tmp->f.mallocd = 0;
   tmp->f.offset = AST_FRIENDLY_OFFSET;
   tmp->f.src = __PRETTY_FUNCTION__;
@@ -226,7 +226,7 @@ lintoalaw_frameout (struct ast_translator_pvt *pvt)
   if (tmp->tail) {
 	  tmp->f.frametype = AST_FRAME_VOICE;
 	  tmp->f.subclass = AST_FORMAT_ALAW;
-	  tmp->f.timelen = tmp->tail / 8;
+	  tmp->f.samples = tmp->tail;
 	  tmp->f.mallocd = 0;
 	  tmp->f.offset = AST_FRIENDLY_OFFSET;
 	  tmp->f.src = __PRETTY_FUNCTION__;
@@ -249,7 +249,7 @@ alawtolin_sample ()
   f.frametype = AST_FRAME_VOICE;
   f.subclass = AST_FORMAT_ALAW;
   f.datalen = sizeof (ulaw_slin_ex);
-  f.timelen = sizeof(ulaw_slin_ex) / 8;
+  f.samples = sizeof(ulaw_slin_ex);
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
@@ -269,7 +269,7 @@ lintoalaw_sample ()
   f.subclass = AST_FORMAT_SLINEAR;
   f.datalen = sizeof (slin_ulaw_ex);
   /* Assume 8000 Hz */
-  f.timelen = sizeof (slin_ulaw_ex) / 16;
+  f.samples = sizeof (slin_ulaw_ex) / 2;
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c
index c84ee2897..6a3ff8054 100755
--- a/codecs/codec_gsm.c
+++ b/codecs/codec_gsm.c
@@ -79,7 +79,7 @@ static struct ast_frame *lintogsm_sample()
 	f.subclass = AST_FORMAT_SLINEAR;
 	f.datalen = sizeof(slin_gsm_ex);
 	/* Assume 8000 Hz */
-	f.timelen = sizeof(slin_gsm_ex)/16;
+	f.samples = sizeof(slin_gsm_ex)/2;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -94,7 +94,7 @@ static struct ast_frame *gsmtolin_sample()
 	f.subclass = AST_FORMAT_GSM;
 	f.datalen = sizeof(gsm_slin_ex);
 	/* All frames are 20 ms long */
-	f.timelen = 20;
+	f.samples = 160;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -112,7 +112,7 @@ static struct ast_frame *gsmtolin_frameout(struct ast_translator_pvt *tmp)
 	tmp->f.subclass = AST_FORMAT_SLINEAR;
 	tmp->f.datalen = tmp->tail * 2;
 	/* Assume 8000 Hz */
-	tmp->f.timelen = tmp->tail / 8;
+	tmp->f.samples = tmp->tail;
 	tmp->f.mallocd = 0;
 	tmp->f.offset = AST_FRIENDLY_OFFSET;
 	tmp->f.src = __PRETTY_FUNCTION__;
@@ -190,7 +190,7 @@ static struct ast_frame *lintogsm_frameout(struct ast_translator_pvt *tmp)
 		x++;
 	}
 	tmp->f.datalen = x * 33;
-	tmp->f.timelen = x * 20;
+	tmp->f.samples = x * 160;
 	return &tmp->f;	
 }
 
diff --git a/codecs/codec_lpc10.c b/codecs/codec_lpc10.c
index a2f6bd911..3ee94326c 100755
--- a/codecs/codec_lpc10.c
+++ b/codecs/codec_lpc10.c
@@ -100,14 +100,11 @@ static struct ast_translator_pvt *lpc10_dec_new()
 static struct ast_frame *lintolpc10_sample()
 {
 	static struct ast_frame f;
-	static int longer = 0;
 	f.frametype = AST_FRAME_VOICE;
 	f.subclass = AST_FORMAT_SLINEAR;
 	f.datalen = sizeof(slin_lpc10_ex);
 	/* Assume 8000 Hz */
-	f.timelen = LPC10_SAMPLES_PER_FRAME/8;
-	f.timelen += longer;
-	longer = 1- longer;
+	f.samples = LPC10_SAMPLES_PER_FRAME;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -123,7 +120,7 @@ static struct ast_frame *lpc10tolin_sample()
 	f.datalen = sizeof(lpc10_slin_ex);
 	/* All frames are 22 ms long (maybe a little more -- why did he choose
 	   LPC10_SAMPLES_PER_FRAME sample frames anyway?? */
-	f.timelen = LPC10_SAMPLES_PER_FRAME/8;
+	f.samples = LPC10_SAMPLES_PER_FRAME;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -141,7 +138,7 @@ static struct ast_frame *lpc10tolin_frameout(struct ast_translator_pvt *tmp)
 	tmp->f.subclass = AST_FORMAT_SLINEAR;
 	tmp->f.datalen = tmp->tail * 2;
 	/* Assume 8000 Hz */
-	tmp->f.timelen = tmp->tail / 8;
+	tmp->f.samples = tmp->tail;
 	tmp->f.mallocd = 0;
 	tmp->f.offset = AST_FRIENDLY_OFFSET;
 	tmp->f.src = __PRETTY_FUNCTION__;
@@ -254,7 +251,7 @@ static struct ast_frame *lintolpc10_frameout(struct ast_translator_pvt *tmp)
 	if (tmp->tail < LPC10_SAMPLES_PER_FRAME)
 		return NULL;
 	/* Start with an empty frame */
-	tmp->f.timelen = 0;
+	tmp->f.samples = 0;
 	tmp->f.datalen = 0;
 	tmp->f.frametype = AST_FRAME_VOICE;
 	tmp->f.subclass = AST_FORMAT_LPC10;
@@ -270,9 +267,7 @@ static struct ast_frame *lintolpc10_frameout(struct ast_translator_pvt *tmp)
 		lpc10_encode(tmpbuf, bits, tmp->lpc10.enc);
 		build_bits(((unsigned char *)tmp->outbuf) + tmp->f.datalen, bits);
 		tmp->f.datalen += LPC10_BYTES_IN_COMPRESSED_FRAME;
-		tmp->f.timelen += 22;
-		/* We alternate between 22 and 23 ms to simulate 22.5 ms */
-		tmp->f.timelen += tmp->longer;
+		tmp->f.samples += LPC10_SAMPLES_PER_FRAME;
 		/* Use one of the two left over bits to record if this is a 22 or 23 ms frame...
 		   important for IAX use */
 		tmp->longer = 1 - tmp->longer;
diff --git a/codecs/codec_mp3_d.c b/codecs/codec_mp3_d.c
index f417453ef..c056e1102 100755
--- a/codecs/codec_mp3_d.c
+++ b/codecs/codec_mp3_d.c
@@ -101,7 +101,7 @@ static struct ast_frame *mp3tolin_sample()
 	f.data = mp3_slin_ex;
 	f.datalen = sizeof(mp3_slin_ex);
 	/* Dunno how long an mp3 frame is -- kinda irrelevant anyway */
-	f.timelen = 30;
+	f.samples = 240;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -118,7 +118,7 @@ static struct ast_frame *mp3tolin_frameout(struct ast_translator_pvt *tmp)
 	tmp->f.subclass = AST_FORMAT_SLINEAR;
 	tmp->f.datalen = tmp->tail * 2;
 	/* Assume 8000 Hz */
-	tmp->f.timelen = tmp->tail / 8;
+	tmp->f.samples = tmp->tail;
 	tmp->f.mallocd = 0;
 	tmp->f.offset = AST_FRIENDLY_OFFSET;
 	tmp->f.src = __PRETTY_FUNCTION__;
diff --git a/codecs/codec_ulaw.c b/codecs/codec_ulaw.c
index 175da76f5..2d0a06155 100755
--- a/codecs/codec_ulaw.c
+++ b/codecs/codec_ulaw.c
@@ -151,7 +151,7 @@ ulawtolin_framein (struct ast_translator_pvt *pvt, struct ast_frame *f)
  *
  * Results:
  *  Converted signals are placed in tmp->f.data, tmp->f.datalen
- *  and tmp->f.timelen are calculated.
+ *  and tmp->f.samples are calculated.
  *
  * Side effects:
  *  None.
@@ -168,7 +168,7 @@ ulawtolin_frameout (struct ast_translator_pvt *pvt)
   tmp->f.frametype = AST_FRAME_VOICE;
   tmp->f.subclass = AST_FORMAT_SLINEAR;
   tmp->f.datalen = tmp->tail *2;
-  tmp->f.timelen = tmp->tail / 8;
+  tmp->f.samples = tmp->tail;
   tmp->f.mallocd = 0;
   tmp->f.offset = AST_FRIENDLY_OFFSET;
   tmp->f.src = __PRETTY_FUNCTION__;
@@ -226,7 +226,7 @@ lintoulaw_frameout (struct ast_translator_pvt *pvt)
   if (tmp->tail) {
 	  tmp->f.frametype = AST_FRAME_VOICE;
 	  tmp->f.subclass = AST_FORMAT_ULAW;
-	  tmp->f.timelen = tmp->tail / 8;
+	  tmp->f.samples = tmp->tail;
 	  tmp->f.mallocd = 0;
 	  tmp->f.offset = AST_FRIENDLY_OFFSET;
 	  tmp->f.src = __PRETTY_FUNCTION__;
@@ -249,7 +249,7 @@ ulawtolin_sample ()
   f.frametype = AST_FRAME_VOICE;
   f.subclass = AST_FORMAT_ULAW;
   f.datalen = sizeof (ulaw_slin_ex);
-  f.timelen = sizeof(ulaw_slin_ex) / 8;
+  f.samples = sizeof(ulaw_slin_ex);
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;
@@ -269,7 +269,7 @@ lintoulaw_sample ()
   f.subclass = AST_FORMAT_SLINEAR;
   f.datalen = sizeof (slin_ulaw_ex);
   /* Assume 8000 Hz */
-  f.timelen = sizeof (slin_ulaw_ex) / 16;
+  f.samples = sizeof (slin_ulaw_ex) / 2;
   f.mallocd = 0;
   f.offset = 0;
   f.src = __PRETTY_FUNCTION__;