Please change the code in hfc_pci.c function hfcpci_clear_fifo_tx :

	bzt->za[MAX_B_FRAMES].z2 = bzt->za[MAX_B_FRAMES].z1;
in
	bzt->za[MAX_B_FRAMES].z2 = bzt->za[MAX_B_FRAMES].z1 -1;


This will solve the problem. Asterisk will function correctly afterwards. 

UPDATE 06-01-05
After getting our Cisco 7960 IP Phones (we use them in SIP mode) we discovered another problem that is troubling Asterisk in combination with ISDN4Linux.
The problem is that we are unable to send outgoing DTMF tones to the PSTN. The fix consists out of the following patch to isdn_tty.c of the Linux kernel:

--- /root/isdn_tty.c	2003-11-05 18:02:49.000000000 +1100
+++ isdn_tty.c	2003-11-05 18:08:46.000000000 +1100
@@ -133,9 +133,9 @@
 			if (info->online) {
 				r = 0;
 #ifdef CONFIG_ISDN_AUDIO
-				isdn_audio_eval_dtmf(info);
-				if ((info->vonline & 1) && (info->emu.vpar[1]))
-					isdn_audio_eval_silence(info);
+				//isdn_audio_eval_dtmf(info);
+				//if ((info->vonline & 1) && (info->emu.vpar[1]))
+				//	isdn_audio_eval_silence(info);
 #endif
 				if ((tty = info->tty)) {
 					if (info->mcr & UART_MCR_RTS) {
@@ -190,10 +190,10 @@
 #ifdef CONFIG_ISDN_AUDIO
 	ifmt = 1;
 	
-	if ((info->vonline) && (!info->emu.vpar[4]))
-		isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt);
-	if ((info->vonline & 1) && (info->emu.vpar[1]))
-		isdn_audio_calc_silence(info, skb->data, skb->len, ifmt);
+	//if ((info->vonline) && (!info->emu.vpar[4]))
+	//	isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt);
+	//if ((info->vonline & 1) && (info->emu.vpar[1]))
+	//	isdn_audio_calc_silence(info, skb->data, skb->len, ifmt);
 #endif
 	if ((info->online < 2)
 #ifdef CONFIG_ISDN_AUDIO

UPDATE 06-01-2005
I discovered today that Skinny IP phones will not hear a ringing tone when calling to another extension using Asterisk. Here is a patch that will fix this problem (quick and dirty):
Apply this to channels/chan_skinny.c :

@@ -1715,14 +1756,17 @@
        }
        switch(ind) {
        case AST_CONTROL_RINGING:
-               if (ast->_state == AST_STATE_RINGING) {
+               ast_verbose(VERBOSE_PREFIX_3 "State AST_CONTROL_RINGINGn");
+               // if (ast->_state == AST_STATE_RINGING) {
+               ast_verbose(VERBOSE_PREFIX_3 "State AST_STATE_RINGINGn");
                        if (!sub->progress) {           
                                transmit_tone(s, SKINNY_ALERT);
                                transmit_callstate(s, l->instance, SKINNY_RINGOU
T, sub->callid);
                                sub->ringing = 1;
+                               ast_verbose(VERBOSE_PREFIX_3 "Started Ringingn"
);
                                break;
                        }
-               }
+               // }


Currently we are running the following patch on Asterisk CVS HEAD.