[asterisk-users] Dropped calls when all DAHDI lines in use

John Kiniston johnkiniston at gmail.com
Tue Oct 9 10:03:26 CDT 2018

You could use GROUP & GROUP_COUNT to track how many channels you are using
before you attempt to dial out and send back a Busy/Congestion/Whatever to
your endpoint when you are at your limit.

On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin <amartin at xes-inc.com> wrote:

> Hello,
> I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x
> analog
> POTS lines coming into my Asterisk server from the phone company.
> Internally, I
> have about 180 SIP clients defined in sip.conf. What appears to be
> happening is
> that if existing calls are consuming all 8 external lines and a new SIP
> client
> attempts to make a call, an existing call gets dropped. The asterisk log
> simply
> shows this as a normal hangup, so I am not able to easily distinguish
> between a
> normal hangup and this type of dropped call. In testing, I am able to get
> a new
> SIP client to report "service unavailable" when all 8 lines are consumed,
> yet
> still drops are reported.
> I have been unable to find any configuration settings pertaining to
> prioritizing
> existing calls over new calls. What else can I look for to attempt to
> debug and
> fix this so that existing calls are not dropped?
> Thanks,
> Andrew
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