[asterisk-users] Sporadic One Way Audio

[Christian Stredicke]

We have seen cases where an IP address conflict caused something like this.

You can take Wireshark traces on the PC (possibly run them in a loop so that you have a pretty long context) and if you have one-way audio be quick to log on to the web interface of the phone and also take a wireshark (PCAP) trace.

There are a couple of tools available that may help to track such problems down: http://manageengine.adventnet.com/products/vqmanager, http://palladion.net, www.networkinstruments.de, and www.voipfuture.com. I know some of them offer a 14-days demo, and it tremendeously helped on of our clients to fix network problems. You can also use SNMP tools to poll if the phone has any blackouts regaring network availbility (see http://wiki.snom.com/SNMP).

Also the phone sends a statistics at the end of each call. Check the BYE message, there is a counter of received and transmitted packets. Those numbers should be roughtly the same.

CS 

-----Ursprüngliche Nachricht-----
Von: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag von Brent Davidson
Gesendet: Freitag, 24. Oktober 2008 18:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Sporadic One Way Audio

I'm having an unusual problem at one of my branch offices.  Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person.  The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a  Rhino R4FXO-EC card.  Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called.  
Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless.

Are there any common issues that might cause this?

Thanks,
Brent Davidson



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