[asterisk-users] Phones cutting out.....again - PLEASE HELP!! !

[Colin Anderson]

So you need a "divide and conquer" strategy here:

1. Is it Asterisk or the WAN? This should be easy enough to test for. Do
call dropouts happen in your datacentre? If not, your Asterisk install is
good. My money's on the 10mbit WAN pipe, and that's what I would be
focussing on.

2. If it's the WAN, is it a connectivity issue or a bandwidth issue? Do a
continous ping from the remote location to your Asterisk server for a day.
You should get NO packets dropped. If you are getting drops, it's a
connectivity issue and you have to look at your SLA to see what your
provider considers good. Otherwise, bandwidth issue.

3. If it's a bandwidth issue, is it your users doing things or is it a
service that is eating bandwidth? If it's a service that is aggregated to a
remote server, like email, then you can use bandwidth management tools like
AstShape or good old tc to severely retard available bandwidth to the
troublesome service. If it's your users, you have to determine what they are
doing. Look at patterns: Does it happen every Tuesday afternoon when you
know Bob from Accounting is running his reports?

4. Sounds like you are running Asterisk --> SIP --> 10mbit WAN --> SIP -->
Phones - which probably is half the issue right there because of no
jitterbuffer. Dig up an old P-3, stick in Trixbox, run it out to your remote
location, and have your Eyebeam clients use *it* instead of your big
Asterisk server for local connectivity. Then tie your P-3 to your big
Asterisk server with IAX. Jitterbuffer + trunking = goodness and your P-3
won't choke under load if you avoid transcoding by using the same codec
end-to-end. Yes it will blow having to maintain two dialplans. But IAX works
frigging great. I use it to aggregate 30 remote locations over the *public*
Internet to my big Asterisk server, and I never get complaints of dropouts,
and in fact I use it extensively myself and IMO it sounds better* than the
local CableCo's VoIP offering, which is a big POS.

5. Regardless of what it actually is, I would have some sort of traffic
shaper at both ends of the WAN pipe. Again, dig up a couple of old P-2 or
P-3's and stick in a bootable Monowall CD, change the default rules to allow
all traffic through, but create a traffic shaping ruleset to give priority
and bandwidth to 5060, 4569, 10000-20000 and dump everything else to a low
priority queue. 

6. I'd run GSM anyway (even though you tried it) because it would eliminate
half your bandwidth consumption. Another variable eliminated.

hth

*By 'sounds better' I mean it sounds like a perfectly normal PSTN call, ALL
THE TIME in s d  of  co s an ly  s nd ng li e  t hs


-----Original Message-----
From: whois wes [mailto:whoiswes at gmail.com]
Sent: Thursday, July 06, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!


Hate to drag this one back up, but....it's happening again.

Overview of architecture:

Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel
1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the
sangoma a104d with onboard echo can.

Server is located in our data center and connected directly to our
cisco 6513 core switch, so we have almost zero latency. The office
having the issues is located several miles away and is connected via a
10Mbit fiber pipe, also low latency. Ping times between remote office
and here are well under 10ms.

T1's are robbed-bit, E&M wink signalling <--- (this may be cause, but
want your input).

Server load is averaging around 20%, plenty of memory, disk space, and
bandwidth available. No QOS running on network. ulaw is the primary
codec.  Server is stable, and there are no extraneous services
running, save mysql and httpd.  Even running a processor intensive
query doesn't trigger the droputs, they happen randomly.

Phones are a mix of Eyebeam 1.5.5 and Eyebeam 1.10 3010n. Both types
of phones are experiencing cutting out of the signal, mainly in the Rx
stream, but occassional in the Tx stream as well. The cutting out was
NOT occurring last night, and the phone server is being rebooted
nightly.  Nothing has changed AT ALL, and the problem has started
occurring again.  If I don't do ANYTHING at all today, there is a 50%
chance that this will NOT occur tomorrow.  In other words, SOMETHING
is causing our phones to drop out, but whatever changes I make seem to
have no effect.  The problem will start and stop seeminly at it's own
whim.

---
Things I have tried:

1.  changed from ulaw to gsm as primary codec - no change
2.  disabled hardware echo can on A104D - no change
3.  moved from asterisk 1.2.4 to 1.2.7.1, recompiled both several
times - no change
4.  have played with gain settings a bit, doesn't seem to make much
difference
---

At this point, i am nearing the end of my rope - i have rebuilt this
machine three times now, and have recompiled the system at least a
dozen times. We have gone from Digium hardware to Sangoma harware and
back again. I have changed every conceivable setting on the phones to
no avail. The problem will randomly disappear, only to come back a few
days later. I can make a change, it seems to have an effect, then
we're back to the same old thing again.

I am in dire need of ANY help anyone can offer, this has been going on
in some form for almost three months.

Thanks for reading,

Wes
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