[asterisk-users] Dropped Calls

[Michael L. Young]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of JR Richardson
> Sent: Tuesday, March 30, 2010 6:55 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [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
> 
> Join the club.  I've experienced the same with various strains on
> 1.4.x above 1.4.21.1 (not an issue with this one that I have seen).
> This issue is truly random and debugging reveals nothing.  I run an
> all SIP environment with same results.  My solution was to downgrade
> to another version or switch to 1.2 or 1.6 depending on what features
> I need for the system.
> 
> Sorry I couldn't be of any help, but I feel your frustration.
> 
> JR
> --
> JR Richardson
> Engineering for the Masses
> 

Is there a chance that you are using Realtime at all?  

I am just curious because I was having problems with dropped calls as well
and just discovered that it appears to be related to the database server.
If for some reason on the database server there is a table lock (which I am
investigating why) asterisk drops any PRI calls and SIP calls.  Everything
looked normal and the error messages never once suggest a problem with the
database server or Realtime.  I was looking everywhere else but at the
Realtime until I stumbled across it.  While doing some backups with FLUSH
READ LOCKS to a slave machine, which I changed asterisk to use a few months
back, I had dropped calls occur.  I later confirmed that asterisk seems to
hang / freeze during that period but once the database server releases the
locks, asterisk continues to function without any problems.  

This started to occur when we had an increase in call volume and an increase
in load on the db server.  I was using Realtime for extensions, sip peers
and CDR.  I had turned off using realtime for CDR (which we don't really use
anyway) and started to use a slave server instead of the master when
performing some maintenance on the master db server.  I left it that way
since I was just using it for extensions and sip peers and that had cleared
it up over the last few months until I ran my backup.

Not sure that helps but it is worth a shot in mentioning to you.

Regards,
Michael Young
(elguero)