[Asterisk-Users] Small office setup/using analog lines w/ Ast erisk

[VaibhaV Sharma]

On Tue, 2005-08-23 at 01:30 -0600, Colin Anderson wrote:

> Good luck and please keep posting so everyone can learn from your
> experience. 

Hmmm... we too moved our telephone system at work to asterisk just a few
days back and since then, I have been following this thread.

A few snippets from my experience (which BTW are very similar to the
original poster of this thread).

Old setup
Avaya-Lucent Partner III with 15+ extensions and 8 voice lines

Hardware:
* Dell PowerEdge 1800 with a single Xeon 3.2 ghz

* 2 SATA Hdds with H/W RAID (RAID mainly because we plan to do a lot of
recording on conference calls + fault tolerance)

* 2 TDM400P cards with 8 incoming pots lines

* About 13 polycom IP600, one IP501 and 2 cisco 7960 phone extensions +
a lot of remote sip clients

* One TDM card is Rev I and the other one is a few months older. Even
the red modules on the cards are a mixture of 3 revisions as we bought
them in installments while testing the setup.

Due to the "short deadlines", we decided to go with Asterisk at home for
now and then later on move to a custom programmed asterisk install.
Running AAH 1.3 on the above hardware (darn! 1.5 is already out).

Problems:

1. Poweredge 1800 only had 2 free PCI-X slots. A quick call to digium
   confirmed that the cards would work just fine on those slots.

2. After the install, users complained about low call volumes on both
sides of the calls. This was quickly fixed by tinkering with the rxgain
and txgain in zapata.conf. But this introduced another problem (point
4).

3. On one of the lines, there was a lot of disturbance. It was a
loud/bad buzzing noise which would be heard randomly on any call through
that line. Called up digium and the support technician suggested
swapping the red modules to see if the particular module was faulty. I
rebooted the machine at the end of the day and things were fine even
without swapping the modules. I don't know what the problem was but it
was only on that particular pots line.

4. Major echo and chit-chat noise issues on random calls. Some of our
users have calls running for an average of 30 - 40 mins and they were
experiencing random echo problems to the point that sometimes they would
leave a voicemail on someone's cell phone and the user would hear only
echoed voice and nothing else.

Changing the txgain made the echo worse so I had to settle for 0.0 as
its value. Tried increasing / decreasing echocancel from the default 128
to 256 or 64/32 but echo got worse both ways. So the only change I could
make was the rxgain so that call volume was better on this side. Tried
changing the echotraining values too but that had no effect on the echo.
At this point, I had rxgain set to 10.5 and txgain set to 0.0.

I figured that the echo was being generated maybe by the phones as a few
users complained of echo on internal calls too. So I looked up polycom's
config file and enabled the echo supression tags. That did not help
much. Less, but still random echo problems.

I found a page somewhere on voip-info.org which mentioned that some
digium card revisions have compatibility issues with some motherboards.
The list specifically mentioned the Dell SC series as being problematic.
The newer Poweredge series seemed to be fine but I was not convinced, so
I called up digium and the technician sent me a standard email with
about 4 - 5 things that might be causing the random echo problems. Those
included -

- IRQ sharing issues. Assign separate IRQs to each digium card.

- IDE drives have DMA mode set to on and that IDE drives are preferred
  instead of SATA / SCSI, specifically because IDE drives can be set to
  use UDMA2.

- IRQ misses are bad and should be minimised for the digium cards. This 
  can be checked using zttest application. Minimum acceptable value is
  99.98%

- Avoid running X-windows on the box and disable frame-buffer by using 
  vga=normal on boot.

- If you still see irq misses, try using acpi=off and/or noapic

- Try disabling hyperthreading if it is enabled

Except the IRQ sharing part, I tried everything in that list and nothing
seemed to change. Still lot of random echo on calls. Changing the IRQ
settings was a bit difficult as I only had 2 free PCI-X (now occupied)
slots for the two digium cards, so changing slots was not an option. I
disabled the USB subsystem, floppy drive and the onboard SATA
controller. Even then, one of the digium cards is sharing the IRQ with
the network controller and I see no way to be able to change that. I am
thinking of calling dell if this situation persists.

Day before yesterday, I took the TDM cards out and swapped a few modules
between themselves. That did not change anything.

Then just yesterday, I found this URL in one of the emails on this
thread - 

http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html

Out of this list, I tried the following just yesterday evening -

Recompile zaptel with 

- MMX enabled

- Enable the AGGRESSIVE_SUPPRESSOR with MARK2

Since this morning, no one has complained of any echo. Looking at that,
I just tried increasing the rxgain to 5.0 and still no echo. However, I
can feel the echo supression algorithm working fulltime to "absorb" my
voice on the phone and not let it echo. I am hopeful that this continues
to work well and I can increase the rxgain further so that call volume
can improve.

I still feel that the transmit volume is too low from our side and thats
because of the polycom phones. Next thing to tinker would be the txgain
in the polycom configs.

My $0.02 in this thread.

--
VaibhaV
http://vsharma.net