<span class="gmail_quote"></span>Hi, <br> I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-
13.diff.gz
(I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT, another TE. My configurations:<br><br>/etc/zaptel.conf:
<br>loadzone=lt<br>defaultzone=lt<br>span=1,1,3,ccs,ami<br>bchan=1-2<br>dchan=3<br>span=2,0,3,ccs,ami<br>bchan=4-5<br>dchan=6<br><br>/etc/asterisk/zapata.conf<br>[channels]<br><br>switchtype = euroisdn<br>; p2mp NT mode
<br>signalling = bri_net_ptmp<br><br>echocancel=yes<br>echotraining = 100<br>echocancelwhenbridged=yes<br><br>immediate=yes<br>group = 1<br>context=from-internal<br>channel => 1-2<br><br>signalling = bri_cpe_ptmp<br>group = 2
<br>context=from-internal<br>channel => 4-5<br><br><br>So I get this from asterisk logs:<br><br>Aug 12 03:08:23 DEBUG[2997] chan_sip.c: build_route: Contact hop: <<a href="mailto:sip:100@192.168.0.1" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
sip:100@192.168.0.1
</a>><br>Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Executing Dial("SIP/100-081c6668", "ZAP/g1/52040004|60") in new stack<br>Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Requested transfer capability: 0x00 - SPEECH
<br>Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Called g1/52040004<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1
<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: disabled echo cancellation on channel 1<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Updated conferencing on 1, with 0 conference users<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1<br>Aug 12 03:08:25 DEBUG[3046] chan_zap.c: disabled echo cancellation on channel 1
<br>Aug 12 03:08:25 VERBOSE[3046] logger.c: -- Hungup 'Zap/1-1'<br>Aug 12 03:08:25 DEBUG[3046] app_dial.c: Exiting with DIALSTATUS=CANCEL.<br><br><br><br>And kernel log:<br>Aug 12 11:01:25 pbx kernel: zaphfc: Card 0 configured for NT mode
<br>Aug 12 11:01:25 pbx kernel: zaphfc: Card 0 configured for master mode<br>Aug 12 11:01:25 pbx kernel: ACPI: PCI Interrupt 0000:04:01.0[A] -> GSI 17 (level, low) -> IRQ 177<br>Aug 12 11:01:25 pbx kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xe01faf00 fifo 0xdaef0000(0x1aef0000) IRQ 177 HZ 250
<br>Aug 12 11:01:25 pbx kernel: zaphfc: Card 1 configured for TE mode<br>Aug 12 11:01:25 pbx kernel: zaphfc: Card 1 configured for master mode<br>Aug 12 11:01:25 pbx kernel: zaphfc: 2 hfc-pci card(s) in this box.<br>Aug 12 11:01:25 pbx kernel: Registered tone zone 22 (Lithuania)
<br><br><br>Aug 12 03:08:25 pbx kernel: zaphfc[1]: b channel buffer underrun: 1, 0
<br>Aug 12 03:08:25 pbx kernel: zaphfc[1]: b channel buffer overflow: 24, 24<br><br><br>Any ideas , how to fix this problem? Thanks<br>