[Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd
behavior (help!)
Gaurav Naik
gnaik at minerva.ece.drexel.edu
Thu Dec 8 17:40:23 MST 2005
I'm having a strange problem with an analog line connected to an
Adtran Channel Bank. It seems as tho, I cannot make outgoing calls
out of the PBX the analog lines are connected to. I'll explain...
The channel bank has a few analog lines (loop start) coming in to the
FXO cards from a Nortel Meridien Option 81c. I have no
administrative/technical control over the Nortel, so I have to
believe that the lines are configured correctly (nor are they willing
to setup a PRI).
As a start, I terminated one of the analog lines into an analog phone
and was able to successfully make and receive calls. Next, I
connected this same line to a X100P and had no problems with Asterisk
(other than disconnect supervision).
Finally, I connected the line to the channel bank (channel 1 of the
T1). The T1 between TE110P and the Channel Bank is up -- no errors.
The TE110P is the master, with the 750 as the slave. I could dial a
4-digit extension number and connect to any station on the Nortel
switch (albeit with some echo, but no static). However, dialing
outside of the Nortel doesn't work.
A '9' has to be dialed in order to "get" an outside line. First, I
started with a SIP phone, and setup my dial plan accordingly. Stuck
a few No-Ops in there and watched on the console. Asterisk was
correctly dialing 91800XXXXXXXX on the FXO port, however, the Nortel
kept ringing extension 1800. Next, I tried adding waits
(w9w1800XXXXXXX), but I still keep getting extension 1800. I even
tried longer and longer waits. I turned on zap debugging from the
asterisk CLI, and could see that DTMF 9 was being sent. Hmm. Again,
with X100P, I could dial outside no problem -- using the SAME dial plan.
In order to make debugging this problem easier, I hooked up the
analog phone to one of the FXS ports, and had Asterisk do a native
bridge of the two ZAP lines. Fine, so now I'm hearing the dialtone
from the Nortel on my analog set. I dial '9', there a slight silence
(500ms or so) and I get a dial tone. Correct behavior. Next, I dial
1800XXXX....extension 1800 is ringing. That didn't work. I cycled
thru about 20 different configurations of zapata.conf and
zaptel.conf. Still doesn't work. I'm beginning to wonder if there
is something wrong with the Adtran.
For a change, I decided to try incoming calls. Asterisk gets the
ring and answers (although it does report something about the line
ringing in the wrong state), however, it cannot decode the DTMF
digits the caller is dialing. I dial 813 and Asterisk thinks I
dialed 83. My dial plan to setup to report an invalid extension and
gives the user another chance to enter the extension. The second try
always works. (I did this about 5 times, with no problem.) Hmm. So
it gets the wrong DTMF digits the first time, every time. I checked
the Digit/Response timeouts...they are set to 5 (for both cases). I
even tried the relaxdtmf directive, but to know avail. Again, I
wasn't having these problems on X100P. The only difference is that
the X100P is running Asterisk v1.0.9 and is in an older machine.
I've gone through every configuration directive possible (disabling
usecallerid, callprogress, echocancel, etc. etc. ). I've even tried
dialing 9 three times. Then I pulled out a multi-meter and made sure
that Tip and Ring weren't reversed (although that shouldn't make a
difference). I checked zttool for IRQ misses and it reported none.
The dialtone sounds fine (no hiss, pops, or static), but after
checking the asterisk/zaptel configuration 25 times, I'm beginning to
think its a wiring problem. Its possible that the 750 isn't grounded
correctly..that is probably my next step.
The relevant portions of my current configuration follow. Any
assistance, or hints and tips for debugging this problem are
appreciated.
Thanks in advance,
--
Gaurav Naik
...apologizing for the long e-mail.
*******
System Config
--
Dell Poweredge 1550 server
Digium TE110P (master clock)
Adtran TA 750 (slave, and two 4-port FXO cards, and 1 4-port FXS card)
Asterisk/Zapata v1.2
RHEL v4.0 Advanced Server
Polycom/Grandstream SIP Phones
a few plain old analog phones
/etc/zaptel.conf
--
span=1,0,0,esf,b8zs
fxsls=1
fxols=9
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf (this is version 34325, the simplest one).
--
signalling=fxo_ls
language=en
context=from-analog-phone
channel => 9
signalling=fxs_ls
language=en
context=from-outside
channel => 1
Adtran TA 750
--
FXO Loop Start (Time Slot 1)
TX Attenuation 0.0
RX Attenuation 0.0
All other FXO ports are disabled. (i've tried it with all the FXO
ports enabled as well).
FXS Loop Start (Time Slot 9)
More information about the asterisk-users
mailing list