@Marrie For one way audio as a debug strategy you can enable RTP debug and see whether you have both way packets flow SENT and GOT.
Regards On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer <li...@jttech.se> wrote: > 2013-05-02 13:19, Marie Fischer skrev: > > 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. > > > > 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. > > > > Voipmonitor.org is great for debugging voip. You can either use only the > sniffer (opensource) and use mysql + the pcap files or you can also buy > the commercial webgui. Either way, it's a great product. > > /Johan > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
