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

Andrew Martin amartin at xes-inc.com
Tue Oct 9 09:21:29 CDT 2018

----- Original Message -----
> From: "John Novack SCII_U" <jnovack at comcast.net>
> To: "Asterisk Users Mailing List, Non-Commercial Discussion" <asterisk-users at lists.digium.com>, "Andrew Martin"
> <amartin at xes-inc.com>
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all DAHDI lines in use

> Have you given any thought to moving to at least a current supported version 13?
> Asterisk 11 has been EOL for some time now
> I doubt you will get a resolution to a version no longer supported.
> Moving to the latest version 13 should be relatively quick and painless, and if
> the issue persists you might find more assistance.
> John Novack
> Andrew Martin 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
> --
> Dog is my Co-Pilot


Thanks for the reply. Yes, I am planning on moving to version 13 but need to find a
solution in the interim. If there are any configuration options that pertain to 
which actions to take with existing calls when new calls come in, I think it is likely
that they would be shared between both versions (and I want to make sure I have the
correct settings when I switch to version 13 too). Can you advise on any tunables
related to handling existing vs new calls?



More information about the asterisk-users mailing list