I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the audio will drop out for between 15 and 20 seconds before suddenly coming back. I've tried running u-Law as the codec and licensed g.729 version 13.0_3.1.7 with exactly the same results. I have tried turning on every logging option I can think of to troubleshoot this but have not been able to find a solution. I'm troubleshooting by remote, so haven't been able to run a wireshark capture yet. pings to the phones from the Asterisk server show no packet loss during the cut-outs. Any ideas? Thanks, *Brent Davidson* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160823/8e66f9c8/attachment.html>
I had this recently... and i bet if you use wireshark/tcpdump youll see a dns lookup for the server's own hostname right before the cutout, and audio again after response is received. quick fix is to add the hosts name and ip to /etc/hosts https://issues.asterisk.org/jira/browse/ASTERISK-26280 On Tue, Aug 23, 2016 at 12:20 PM, Brent Davidson < brent at texascountrytitle.com> wrote:> I'm having an issue with some Snom 300s on a server running Asterisk > version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is *NO NAT* > involved. Phones and server are plugged into the same network switch, all > on the same IP range. The server is running a Wildcard AEX410 analog card > with 2 FXO modules receiving incoming analog lines. > > Occasionally, in the middle of a call, the audio will drop out for between > 15 and 20 seconds before suddenly coming back. I've tried running u-Law as > the codec and licensed g.729 version 13.0_3.1.7 with exactly the same > results. I have tried turning on every logging option I can think of to > troubleshoot this but have not been able to find a solution. I'm > troubleshooting by remote, so haven't been able to run a wireshark capture > yet. > > pings to the phones from the Asterisk server show no packet loss during > the cut-outs. > > Any ideas? > > Thanks, > *Brent Davidson* > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160823/043e07f2/attachment.html>
I dont know the rules of the mailing list, but im curious if this fixed it for you. I'm just that kind of person. do you have any confirmation? On Tue, Aug 23, 2016 at 12:27 PM, eli vaughan <edong23 at gmail.com> wrote:> I had this recently... and i bet if you use wireshark/tcpdump youll see a > dns lookup for the server's own hostname right before the cutout, and audio > again after response is received. quick fix is to add the hosts name and ip > to /etc/hosts > > https://issues.asterisk.org/jira/browse/ASTERISK-26280 > > On Tue, Aug 23, 2016 at 12:20 PM, Brent Davidson < > brent at texascountrytitle.com> wrote: > >> I'm having an issue with some Snom 300s on a server running Asterisk >> version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is *NO NAT* >> involved. Phones and server are plugged into the same network switch, all >> on the same IP range. The server is running a Wildcard AEX410 analog card >> with 2 FXO modules receiving incoming analog lines. >> >> Occasionally, in the middle of a call, the audio will drop out for >> between 15 and 20 seconds before suddenly coming back. I've tried running >> u-Law as the codec and licensed g.729 version 13.0_3.1.7 with exactly the >> same results. I have tried turning on every logging option I can think of >> to troubleshoot this but have not been able to find a solution. I'm >> troubleshooting by remote, so haven't been able to run a wireshark capture >> yet. >> >> pings to the phones from the Asterisk server show no packet loss during >> the cut-outs. >> >> Any ideas? >> >> Thanks, >> *Brent Davidson* >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 >> http://www.asterisk.org/community/astricon-user-conference >> >> 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160828/3defa1fb/attachment.html>