[asterisk-users] Dropped Calls

[Danny Nicholas]

A few thoughts;
1. I assume that the * servers aren't on dedicated networks;  Do the dropped
or one-way calls occur during high-traffic times or are they concurrent with
large downloads?  In my shop, we had to get a router that would prioritize
voice traffic or we would be dead in the water during client file
transmissions.
2. Don't know about the SNOM or GS phones, but my Polycom phones let you
establish higher packet priorities for voice traffic as well.
3. Have you been able to do a "top" during one of these failures?  Could be
a memory leak that comes up randomly.
4. Looking at the startup logs, are the cards having to retry several times
to get an IRQ?  Digium cards IME can conflict with the Hard Drive (SCSI)
controller, causing problems during heavy I/O periods.
Hope this helps.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brent Davidson
Sent: Tuesday, March 30, 2010 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped Calls

I've written about this issue several times, but have not yet found any 
solution to it.  I am using asterisk 1.4.21.2 and zaptel 1.4.12.  Phones 
are primarily Snom 300's but I also have a couple of headset phones 
connected to Grandstream HT286 SIP adapters.  I have 8 offices, each has 
it's own asterisk server all running the same versions of asterisk and 
Zaptel.  Only difference is that one office uses a Digium TDM 8-port 
card and the other branches use 4-port Rhino cards with only 2 ports in 
use.  What happens is that periodically we will be in a call and the 
call will just drop.  It's usually within the first couple of minutes of 
the call.  The calls can be either incoming or outgoing.  The phenomenon 
affects both the Snoms and the Grandstreams.  Along with the dropped 
call issue, we periodically have a problem where a person we call or a 
person that calls in cannot hear the person in the our office, but the 
person in our office can hear the remote person fine.

All of the phones are on the same physical network as the asterisk 
server.  There is no NAT, no Firewall, VLAN, etc. between the phones and 
the server.   I have tried running sip debugs on the calls, but on the 
off chance that my logs catch either a drop or a one-way audio, the sip 
debug looks like just a normal call.

Is there any setting that might cause both one-way audio and dropped calls?

Thanks,
Brent Davidson

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