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brent at texascountryt... Guest
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Posted: Tue Aug 23, 2016 12:21 pm Post subject: [asterisk-users] Audio cut-outs |
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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 |
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edong23 at gmail.com Guest
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Posted: Tue Aug 23, 2016 12:27 pm Post subject: [asterisk-users] Audio cut-outs |
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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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote: | 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
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edong23 at gmail.com Guest
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Posted: Sun Aug 28, 2016 1:52 pm Post subject: [asterisk-users] Audio cut-outs |
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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@gmail.com (edong23@gmail.com)> wrote:
Quote: | 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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote: | 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
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