[Asterisk-Users] my asterisk experience (long)

[Sean Rodger]

I thought I'd post my experiences for the benefit of anyone else who may be
at the point I was when I first started with asterisk.

I have 2 incoming analog lines (north eastern U.S., Verizon) where one is
set to ring if the first is busy.
I bought a bare-bones system from abs-pc with the following components:

POWER SUPPLY 450W ALLIED ATX450P4 R(41)
MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard)
CPU AMD|2500/333 ATHLON XP BARTON R(Standard)
DDRAM 256M|DDR333 PC-2700 -K %(Standard)
HD 40GB|WD 7200RPM 8MB   WD400JB%(70)
VGA ASUS|V8170MAGICII/T 64M MX440SE(58)
CD ROM 56X|AOPEN CD-956 RTL(22)

I also bought 2 X100P's and 1 TDM400P from Digium, and installed them in the
above system.

I installed RedHat 9 onto the PC.  During the RH install, I selected the
"server" install, and tried to weed out most of the packages that I didn't
need.  I'm no Linux expert, but I didn't want a lot of stuff running on my
server. IMO simple is better (and more secure).  Along these same lines, I
ran the RH command 'setup' and turned off all of the services that I didn't
need.  I would do the same with the kernel, but I'm not that Linux savvy
yet.

Setting up Linux, installing Asterisk, and writing some basic conf files
took about 2 weeks in my spare time. Most of that time was spent learning
about asterisk, and what I needed to include in my conf files.  My initial
conf files were mostly adaptations of others that I found around on the net.

I bought two radioshack single line phones (one was cordless), plugged them
into the TDM400P.  After getting the drivers loaded, and asterisk running, I
ran into my first problem.  I've covered this problem extensively in earlier
posts (subject: "TDM400P??"), so I will just briefly mention it here.  The
Pro-SLIC modules were resetting on hook transitions.  Its like they were not
getting enough power.
After much debugging, and work with Digium, the problem could not be solved.
I sent the card back to Digium, and they sent me a new one.  The new card
behaved the same way.  Mark edited the driver on my machine to prevent the
module reset from crashing the wcfxo driver, but the problem was not solved.
Eventually I came to accept that the card simply did not work with my
motherboard, an ASUS A7n8X-Deluxe.  Digium refunded my money for the card,
and I returned to the drawing board.

I bought a Grandstream 101, then I bought 2 more.  I also got a Cisco
ATA186.   I had looked into using the ATA186 with asterisk, and it looked
like I could get it to work.  When I got it, I realized that It didn't have
the same firmware as I thought it would.  In fact, as it was, I couldn't get
it to work with asterisk at all.  I tried to get a firmware update from the
Cisco website.  Their website is ridiculously complex and annoying.  In the
end, though the web site didn't tell me this explicitly, I found that they
would not let me download a firmware upgrade.  Luckily I was able
successfully navigate their huge and annoying phone system to reach an
engineer who was nice enough to email me the SIP firmware upgrade "as a
courtesy".  After I loaded that firmware the Cisco ATA186 has worked good.

The motherboard I am using has 2 Ethernet ports, but RH9 only recognizes
one.  I downloaded a Linux driver from NVIDIA, and had to manually edit the
/etc/sysconfig files; redhat's config menus can't handle 2 Ethernet ports
apparently.  I set a DHCP server to run on the second Ethernet port, and
also set up a NTP server for the grandstream phones' time display.  I did
not set up a route between the two ports.  This gives me a separate isolated
network for my BT-101's and the ATA186.

I recorded audio using a regular PC mic, and Goldwave.  Goldwave is nice as
it lets you edit wav files, equalizing volumes, and applying filters.  I
converted the files from wav to gsm using Sox.

After I got all of this set up I began testing after-hours in the office.
The echo problem immediately became obvious.
Everything else seemed to work good.  I set the grandstream phones to use
SIP-info for signaling, and spent some time massaging my conf files.
   After activating the Aggressive Suppressor option in the zaptel makefile,
and recompiling the zaptel driver, the echo problem was greatly reduced on
all but one grandstream phone.  I noticed that one phone had older firmware.
I set up a tftp server, and updated the BT-101's firmware.  The firmware
upgrade seemed to fix the remaining echo on that one phone.

The echo is still audible as the occasional chirp or crackle, but it is now
at a tolerable level.  There is the additional problem of regular speech
audio occasionally getting suppressed when both parties start talking at the
same time.  That is not a bad problem as it doesn't happen often, and
quickly fixes itself.

The current system is working good, except for the above mentioned problems
with audio.  The Grandstream phones' function buttons integrate nicely with
asterisk.  All of them seem to work.  I loaded some nice (and hopefully
legal) tunes for the musiconhold (had to install mpg123), and that works
great.

I haven't really had any experience with other PBX's, but after all of the
work I put in, I am happy with the results.  And I don't think anyone at my
company suspects my often irrational commitment to open-source.