[Asterisk-Users] FXO Problems

[Mike Coakley]

I have 2 Digium 4 port FXO cards in my system. The system is a P4 
2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1  for 
storage - whitebox - running RedHat 9. With pretty much any CVS HEAD 
version we are getting, what I will call, "phantom" calls on some 
lines. What I mean by a phantom call is that the line will ring, 
Asterisk will log that the Zap channel has been answered, the context 
will call all of our SIP phones (which works fine) but when you pickup 
the handset you get a dial tone. If you just sit there and listen 
Asterisk will log that the Zap channel has hung up (after about 5 
seconds) and the SIP phone goes busy. I am getting a "reverse polarity" 
message on the console of the Asterisk system (not in the Asterisk 
console but on the monitor attached to the hardware). But I'm not sure 
that is even related yet as today is the first time I saw this message.

I have played with the busydetect and callprogress settings but nothing 
has helped. (So I have left them off for now. With each new CVS HEAD I 
download I try all the settings again but they have not changed the 
situation.) I should also say that this problem did not exist before I 
switched over to the Asterisk system. The key system we had before did 
not exhibit this problem. SO I am assuming it is (a) the new wiring we 
put in place to get to the Asterisk FXO cards (but I have replaced that 
during my diagnosis already), (b) the way that the Digium cards are 
handling the teclo terminations, or (c) my zaptel.conf or zapata.conf 
are somehow wrong.

Here are my configs:

ZAPTEL.CONF
#
# Zaptel Configuration File
#
fxsks=1-8
loadzone = us
defaultzone=us

ZAPATA.CONF
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]
;
; All channel defaults
;
language=en
context=pstn_inbound
signalling=fxs_ks
musiconhold=default

relaxdtmf=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=5.0
txgain=0.0
busydetect=no
busycount=6
callprogress=no

callgroup=1
pickupgroup=1

;We have this disabled for now since we don't have Caller ID on our 
lines
usecallerid=no
useincomingcalleridonzaptransfer=yes

;We group the following lines into a single group for
; pooled outbound calls
group=1

;Define our incoming/outgoing lines
callerid="MB Main - Line 1" <(973) 252-xxxx>
channel => 1
callerid="MB - Line 2" <(973) 252-xxxx>
channel => 2
callerid="MB - Line 3" <(973) 252-xxxx>
channel => 3
callerid="MB - Line 4" <(973) 252-xxxx>
channel => 4
callerid="MB - Line 5" <(973) 252-xxxx>
channel => 5
callerid="MB - Line 6" <(973) 252-xxxx>
context=pstn_fax_inbound
channel => 6

I'm sure there are some telco gurus out there that can tell me how to 
test the incoming telco lines to determine where my problem is. My LEC 
is Verizon in New Jersey for those who care.

Some more background: I have had other problems on the line. I had a 
beeping or blip on the line towards the caller (not the receiving SIP 
side) that sounded like the blips you would get if you were recording 
the call. Reading some other posts it sounded like another problem 
someone else was having and giving the Digium FXO cards a higher PCI 
latency timing helped that situation. Also I had echo on the lines and 
no settings in the configs helped at all. I had to uncomment the 
AGRESSIVE_SUPRESSOR for the MARK2 echo cancellation algorithm to get my 
echo to go away (thanks Mark from Digium for that one).

Thanks,

Mike