<div dir="ltr">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.</div><br><div class="gmail_quote"><div dir="ltr">On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin <<a href="mailto:amartin@xes-inc.com">amartin@xes-inc.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog <br>
POTS lines coming into my Asterisk server from the phone company. Internally, I<br>
have about 180 SIP clients defined in sip.conf. What appears to be happening is<br>
that if existing calls are consuming all 8 external lines and a new SIP client<br>
attempts to make a call, an existing call gets dropped. The asterisk log simply<br>
shows this as a normal hangup, so I am not able to easily distinguish between a<br>
normal hangup and this type of dropped call. In testing, I am able to get a new<br>
SIP client to report "service unavailable" when all 8 lines are consumed, yet<br>
still drops are reported.<br>
<br>
I have been unable to find any configuration settings pertaining to prioritizing<br>
existing calls over new calls. What else can I look for to attempt to debug and<br>
fix this so that existing calls are not dropped?<br>
<br>
Thanks,<br>
<br>
Andrew<br>
<br>
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