[asterisk-users] Dropped Calls

[JR Richardson]

> 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