asterisk users - Oct 2018 - Dropped calls when all DAHDI lines in use

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

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

----- 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
John,

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?

Thanks,

Andrew

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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