[asterisk-users] debug strategy for one-way audio calls

[Olivier]

Le 2 mai 2013 13:23, "Marie Fischer" <marie at vtl.ee> a écrit :
>
> Hello everybody,
>
> from time to time, we get so-called simplex / one-way audio calls, where
one party cannot hear the other. The only thing in common is that is does
happen with calls via SIP trunk, not ISDN and not internal calls. Nothing
strange in verbose and SIP logs. Could even be some weird intermittent
firewall issue I guess.
>
Which audio flow is missing ? Inbound ?

I suppose it should be easier to automatically detect missing inbound audio.

> Apart from logging all traffic 24/7 via tcpdump (not really convenient),
can you give me some ideas how to debug this kind of issue?
>
> Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.
>
> --
> Thanks,
>
> marie
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> 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/20130504/23447174/attachment.htm>