[Asterisk-Users] Voicetronix Audio Problems when making two or more simultanoues calls

[Peter Zion]

Hi David,

I've been working with an * setup with a VoiceTronix 4-port FXO card and
I had similar problems with detecting dialtone.  I had to chan_vpb.c to
use a VPB driver call progress dial command, and to change the dialtone
dectection (I'm in North America) -- please see patch against version
0.7.1 below.  I have not submitted this patch upstream because it's a
workaround that works for my application but leaves lots of problems
unfixed, and possibly creates other problems, so use at your own risk!

I have not yet worked with more than one line with this card so I cannot
say whether I have the other problem you mentioned.

In general, my experience using this card with Asterisk has been
negative.  I hope that Digium will be releasing a multi-port FXO card
soon.

Peter Zion


--BEGIN PATCH--
diff -rpuN ../.build_orig/asterisk/channels/chan_vpb.c
asterisk/channels/chan_vpb.c
--- ../.build_orig/asterisk/channels/chan_vpb.c	2004-01-28
12:36:42.000000000 -0500
+++ asterisk/channels/chan_vpb.c	2004-01-28 12:36:35.000000000 -0500
@@ -280,17 +280,21 @@ static inline int monitor_handle_owned(s
 	  break;
 
      case VPB_CALLEND:
-	  if (e->data == VPB_CALL_CONNECTED)
+	  if (e->data == VPB_CALL_CONNECTED ||
+                  e->data == VPB_CALL_NO_RING_BACK)
 	       f.subclass = AST_CONTROL_ANSWER;
-	  else if (e->data == VPB_CALL_NO_DIAL_TONE ||
-		   e->data == VPB_CALL_NO_RING_BACK)
+	  else if (e->data == VPB_CALL_NO_DIAL_TONE)
 	       f.subclass =  AST_CONTROL_CONGESTION;
 	  else if (e->data == VPB_CALL_NO_ANSWER ||
 		   e->data == VPB_CALL_BUSY)
 	       f.subclass = AST_CONTROL_BUSY;
 	  else if (e->data  == VPB_CALL_DISCONNECTED) 
 	       f.subclass = AST_CONTROL_HANGUP;
-	  break;
+
+    	  if (f.subclass != AST_CONTROL_ANSWER)
+               vpb_sethook_sync(p->handle, VPB_ONHOOK);
+
+ 	  break;
 
      case VPB_STATION_OFFHOOK:
 	   f.subclass = AST_CONTROL_ANSWER;
@@ -459,8 +463,12 @@ static void *do_monitor(void *unused)
 	  ast_mutex_lock(&monlock),
 	       ast_mutex_lock(&iflock); {
 	       struct vpb_pvt *p = iflist; /* Find the pvt structure */	      
+               int len;
 		
 	       vpb_translate_event(&e, str);
+               len = strlen(str);
+               if (len > 0 && str[len-1] == '\n')
+                    str[len-1] = '\0';
 	       
 	       if (e.type == VPB_NULL_EVENT) 
 		    goto done; /* Nothing to do, just a wakeup call.*/
@@ -546,7 +554,8 @@ struct vpb_pvt *mkif(int board, int chan
 {
      struct vpb_pvt *tmp;
 
-
+     VPB_DETECT tone;
+     
      tmp = (struct vpb_pvt *)calloc(1, sizeof *tmp);
 
      if (!tmp)
@@ -560,6 +569,13 @@ struct vpb_pvt *mkif(int board, int chan
 	  free(tmp);
 	  return NULL;
      }
+     
+     vpb_gettonedet(tmp->handle, VPB_DIAL, &tone);
+     tone.freq1 = 400;
+     tone.bandwidth1 = 140;
+     tone.glitch = 100;
+     tone.stran[1].tfire = 250;
+     vpb_settonedet(tmp->handle, &tone);
           
      if (echocancel) {
 	  if (option_verbose > 4)
@@ -712,11 +728,15 @@ static int vpb_call(struct ast_channel *
 
 	  vpb_sethook_sync(p->handle,VPB_OFFHOOK);
 	  
-	  res = vpb_dial_async(p->handle, s);
+	  res = vpb_call_async(p->handle, s);
 
 	  if (res != VPB_OK) {
 	    ast_log(LOG_DEBUG, "Call on %s to %s failed: %s\n", 
 		    ast->name, dest, vpb_strerror(res));	      
+		    
+	    vpb_sethook_sync(p->handle, VPB_ONHOOK);
+      	    ast_setstate(ast, AST_STATE_DOWN);
+
 	    res = -1;
 	  } else 
 	    res = 0;
@@ -733,8 +753,6 @@ static int vpb_call(struct ast_channel *
 	      vpb_timer_start(p->timer);
 	 }
 	 p->calling = 1;
-	ast_setstate(ast, AST_STATE_RINGING);
-	ast_queue_control(ast,AST_CONTROL_RINGING, 0);		
     }
 
     return res;
--END PATCH--

On Tue, 2004-02-03 at 00:11, David Liu wrote:
> Hi there,
>  
> Besides the problem of Voicetronix dialing too early before the
> carrier gives a dial tone, there also appears to be issues with the
> audio quality when more than 1 channel is utilized.
>  
> From an experiment, whenever 1 channel is occupied (i.e. outbound call
> in progress) then getting a dial tone on an available channel will
> take sometime.  What this means is, once you get a dial tone after
> pressing 9, the channel is bridged with the SIP phone, then you wait
> about 4 to 5 seconds and you will hear a dial tone from the carrier. 
> The dial tone is very choppy and most of the time we can't even dial
> out because the quality is so bad that the DTMF is not sent out at
> all.  
>  
> It is definitely not the individual line as this scenerio can happen
> at any channel.
>  
> Anybody had similiar problems?  
>  
> David