[asterisk-users] FXO- no dial tone- no call progressing

[Ayman Hendawy]

Dear all,
 I connected a normal phone line to the FXO port but the call is not being
processed. The following is the output to asterisk console when I dial 9150
"9 is the prefix I configured and 150 is a local service in to know the
current time"
*CLI>     -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new
stack
    -- Called
1/150
    -- Zap/1-1 answered SIP/4444-d365

Here are some more details to help in troubleshooting the problem
Dmesg Output is as following:
Zapata Telephony Interface Registered on major
196
Code test: code function addr =
0x004894f4
iRxBuffer1 =
0xff803e58
iTxBuffer1 =
0xff803ed8
ISR installed
OK
port: 1 port_type: O        indicate that asterisk is detect  two FXO
port: 2 port_type:
O
port: 3 port_type:
-
port: 4 port_type:
-
port: 5 port_type:
-
port: 6 port_type:
-
port: 7 port_type:
-
port: 8 port_type:
-
Testing for
DAA...
  VoiceDAA System:
04
  ISO-Cap is now up, line side: 03 rev
06
Module 0: Installed -- AUTO FXO (FCC
mode)
Testing for
DAA...
  VoiceDAA System:
04
  ISO-Cap is now up, line side: 03 rev
06
Module 1: Installed -- AUTO FXO (FCC
mode)
Found: Blackfin STAMP (8
modules)
wcfxs_init_ok =
1
Registered tone zone 0 (United States / North
America)
73318 Polarity reversed (0 -> 1)

I hashed the following lines in zapata.conf corresponding to the FXS ports
which are not installed:

;signalling=fxo_ks
;group=2
;context=internal
;channel=> 3-4

I added the following line to the dial plan under the internal context in
the extensions.conf file to enable routing to the FXO port at channel 1.

exten => _9.,1,Dial(Zap/1/${EXTEN:1})

I added the following section to sip.conf for my IP phone

[4444]
type=friend
host=dynamic
user=4444
secret=1234
disallow=all
allow=ulaw
context=internal

Here are the output of some asterisk commands which might also be useful:

*CLI> zap show
channels
   Chan Extension  Context         Language
MusicOnHold
 pseudo
incoming
      1
incoming
      2            incoming

*CLI> zap show
cadences
r1:
125,125,2000,4000
r2:
250,250,500,1000,250,250,500,4000
r3:
125,125,125,125,125,4000
r4:
1000,500,2500,5000
*CLI> zap show channel
1
Channel:
1
File Descriptor:
12
Span:
1
Extension:

Dialing: no                                 why?

Context:
incoming
Caller ID
string:
Destroy:
0
InAlarm:
0
Signalling Type: FXS
Kewlstart
Owner:
Zap/1-1
Real:
Zap/1-1
Callwait:
<None>
Threeway:
<None>
Confno:
-1
Propagated Conference:
-1
Real in conference:
0
DSP:
yes
Relax DTMF:
no
Dialing/CallwaitCAS:
0/0
Default law:
ulaw
Fax Handled:
no
Pulse phone:
no
Echo Cancellation: 128 taps, currently
ON
DSP cycles last: 752066 worst: 830646 average: 741569 sample:
732930
Actual Confinfo: Num/0,
Mode/0x0000
Actual Confmute:
No
Actual Hookstate: Onhook

 I can measure 52V across the legs of the fxo module. Also when I plug the
cable I receive "Ring on 1/1" message on the console  and when I remove the
cable I receive "No Ring on 1/1". so the PSTN line is sensed by asterisk.

When I try to make a call from my sip phone "4444" on zap 1 port it gave (no
dial tone).


*CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack
-- Called 1/150
-- Zap/1-1 answered SIP/4444-d365

Sometimes it hangs up immediately after answered message. Sometimes it lasts
until I hangup SIP session.

Here is the output of Dmesg during this test.

RING on 1/1!
136232 Polarity reversed (-1 -> 1)
NO BATTERY on 1/1!
BATTERY on 1/1 (-)!
NO BATTERY on 1/1!
BATTERY on 1/1 (-)!
NO BATTERY on 1/1!
NO RING on 1/1!
142815 Polarity reversed (1 -> -1)
BATTERY on 1/1 (+)!
RING on 1/1!
149257 Polarity reversed (-1 -> 1)
NO BATTERY on 1/1!
BATTERY on 1/1 (-)!

I wanted to see the voltage variations on the line during the call to
justify this behavior. I measured the voltage during my test and there was
no variation in voltage across the fxo module legs at all. Is it expected
for the voltage to go down when the call start (off hook). It was 52 and
remains 52 during all tests.

I really don't know why "polarity reversed" and "NO Battery on 1/1" messages
are generated during the call.

When I make incoming call. There is no messages at all on asterisk console
or driver messages on dmesg.

I tried to avoid the hang up because this false polarity reversed signal. I
put the following in zapata.conf. Now it stopped hanging up. But still the
same no dial tone and no call progress.

hanguponpolarityswitch=no

I suspect about the clock  that supplied to the FXO card it is 2.048MHZ
(E1),however I use USA zone configuration and USA work with standard T1
(1.544MHZ).

am i right?
Any hints ?

Thanks in advance.
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