asterisk users - May 2004 - Digital Line Distortion

Firstly, the problem...

Ever since I installed and setup asterisk, I have had various problems,
initially it was echo caused by the ISDN (isdn4linux) card I was using.
So, I upgraded to the X101P from digium. I still had echo, so I figured
it was also caused by the ATA186 (cisco) I was using. So, I upgraded
again to the TDM40B quad FXS card. This solved pretty much all my
problems, except eventually, I needed more incoming lines. So, again, I
upgraded to a digital line (10 channel PRI/E1) and purchased the brand
new TE405p from digium... Now, eventually I got this working properly,
for incoming and outbound calls, I have incoming callerid working,
etc...

However, ever since I did this, I continually get complaints from people
about how terrible my phone lines are. Not *everyone* complains, but
most people do....

Also, I hear the same problem when calls are 'diverted' to my mobile
phone. ie, call arrives, and is then connected to a second channel back
to my mobile.

I still have my X100p installed, and don't have this problem with calls
between the x100p and the tdm40b, only with calls between the te405p and
the tdm40b and calls with both legs on the te405p.

Here are some dumps from my configs/system/etc
asterisk*CLI> show version
Asterisk CVS-04/13/04-18:00:23 built by root@asterisk on a i686 running
Linux

[root@asterisk]: ~# cat /proc/cpuinfo
processor       : 0
vendor_id       : AuthenticAMD
cpu family      : 6
model           : 8
model name      : AMD Athlon(tm)
stepping        : 1
cpu MHz         : 1161.462
cache size      : 256 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 1
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
mca cmov pat pse36 mmx fxsr sse syscall mmxext 3dnowext 3dnow
bogomips        : 2313.42

[root@asterisk]: ~# lspci
00:00.0 Host bridge: VIA Technologies, Inc.: Unknown device 3116
00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP]
00:05.0 Communication controller: Xilinx, Inc.: Unknown device 0314 (rev
01)
00:06.0 Communication controller: Tiger Jet Network Inc. Model 300 128k
00:07.0 Network controller: Tiger Jet Network Inc. Model 300 128k
00:0e.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C (rev 10)
00:11.0 ISA bridge: VIA Technologies, Inc. VT8233A ISA Bridge
00:11.1 IDE interface: VIA Technologies, Inc. Bus Master IDE (rev 06)
01:00.0 VGA compatible controller: S3 Inc.: Unknown device 8d04

[root@asterisk]: ~# cat /proc/interrupts
           CPU0
  0:  683565744    IO-APIC-edge  timer
  1:        874    IO-APIC-edge  keyboard
  2:          0          XT-PIC  cascade
  8:          1    IO-APIC-edge  rtc
 14:    4972250    IO-APIC-edge  ide0
 15:         15    IO-APIC-edge  ide1
 16: 2539525986   IO-APIC-level  t4xxp
 17: 2543170795   IO-APIC-level  eth0, wcfxo
 18: 2539516334   IO-APIC-level  wctdm
NMI:          0
LOC:  683553571
ERR:          0
MIS:          0

Now, some asterisk config files:
[root@asterisk]: ~# cat /etc/zaptel.conf |grep -v ^\#
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
unused=11-15,17-31
dchan=16
fxsls=125
fxoks=126
fxoks=127
fxoks=128
fxoks=129
loadzone = au
defaultzone=au

[root@asterisk]: ~# cat /etc/asterisk/zapata.conf |egrep -v ^$\|^\;
[channels]
context=default
signalling=fxo_ls
usecallerid=no
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
busydetect=no
pridialplan=local
nationalprefix=0
internationalprefix=0011
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
callerid=asreceived
adsi=no
busydetect=no
callprogress=no
switchtype => euroisdn
signalling => pri_cpe
callgroup => 1
group => 2
immediate => no
context => remote
channel => 1-10
immediate => yes
usecallerid=no
callerid=no
group => 10
signalling => fxs_ls
channel => 125
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
context => inside
immediate = no
signalling => fxo_ks
pickupgroup => 1
callgroup => 1
group => 3
callerid="Adam" <601>
mailbox=601
channel => 126
callerid="Doris" <600>
mailbox=600
channel => 127
callerid="Sales" <603>
mailbox=603
channel => 128
callerid="Technical" <602>
mailbox=602
channel => 129

So, you can see the te405p, followed by the x101p and finally the tdm40b

If anyone can help me resolve this terribly annoying problem, I'd be
most appreciative...

I can't think of any other information which is helpful, but please let
me know if you think something else will clearly show the problem (like
a pri debug etc...)

Regards,
Adam

Damn, I forgot to describe the actual problem. Basically as someone I
spoke to today described it, it sounds like you have one of those new
digital pbx systems... In more detail, when he spoke, he heard his voice
come back, but distorted. The louder the sound he made, the louder he
heard himself (distorted).

At all times, if I am on the tdm40b side, I hear 100% perfect audio
quality in both directions. (Which is bad, because now the customer gets
the bad sound, before it was just the staff...)

Regards,
Adam

On Mon, 2004-05-03 at 21:28, Adam Goryachev wrote:
> Firstly, the problem...
> 
> Ever since I installed and setup asterisk, I have had various problems,
> initially it was echo caused by the ISDN (isdn4linux) card I was using.
Now I'll do the thing most people forget about...

[SNIP] the rest of the quoted text!

Regards,
Adam

On Mon, May 03, 2004 at 09:28:56PM +1000, Adam Goryachev
wrote:
> Firstly, the problem...
> 
> Ever since I installed and setup asterisk, I have had various problems,
> initially it was echo caused by the ISDN (isdn4linux) card I was using.
> So, I upgraded to the X101P from digium. I still had echo, so I figured
> it was also caused by the ATA186 (cisco) I was using. So, I upgraded
> again to the TDM40B quad FXS card. This solved pretty much all my
> problems, except eventually, I needed more incoming lines. So, again, I
> upgraded to a digital line (10 channel PRI/E1) and purchased the brand
> new TE405p from digium... Now, eventually I got this working properly,
> for incoming and outbound calls, I have incoming callerid working,
> etc...
> 
> However, ever since I did this, I continually get complaints from people
> about how terrible my phone lines are. Not *everyone* complains, but
> most people do....
> 
We did face what may be the same problem here.

The problem came from the fact that on some motherboards (well, *most*
motherboards, as far as I tested), the TE405P has a problem which
makes it send every one in 8 (or was it 16?) bytes as 0xFF (instead of
whatever the U/A-law value may have been).

On the RX side of things, it was always perfect, thus when connecting to
a local IP phone we heard a nice sound, but the remote party always
had a quite garbled output.

You can check it quite easily :
- plug a crossover cable between two ports
- do not start Asterisk (but load and ztcfg everything)
- cat /dev/zap/span1/1 on one terminal
- ls >/dev/zap/span2/1 on another terminal (provided that spans 1 and
  2 are connected together)
- if everything works well, you should have a perfect output for your
  ls on the "cat" terminal. Otherwise, try hexdump and watch the
  columns with FF.

It was solved by using a PCI 2.2 compliant motherboard (i865
based). It's quite an odd behaviour, and it's still not clear to me
why it happens. I initialiy thought it could be solved by fixing the
FPGA VHDL, but I'm not an expert in that field.

--
Nicolas Bougues
Axialys Interactive