----- Original Message ----- > From: "John Novack SCII_U" <[email protected]> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > <[email protected]>, "Andrew Martin" > <[email protected]> > 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
