On Fri, 2011-05-27 at 10:31 +0200, Mark Scholten wrote: > Hello, > > We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. > > SIP clients (all behind NAT at different locations, so not a single NAT > solution is used): > - x-lite > - linksys pap2t > - polycom kirk (multiple type numbers) > - polycom (multiple type numbers, hardware phones) > > Our Asterisk servers stays in between (some calls are recorded). Asterisk is > running on a physical server (no virtual server software) with "old" > hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for > buffers). We use a MySQL backend (CDR records are stored in it and SIP users > are stored in a MySQL database). > > We use a SIP provider with a trunk for outgoing and incoming calls, this is > also an Asterisk server if I'm correct. We currently do around 1000 calls a > week and max. do 10 calls at the same time. The Asterisk server is not > behind a NAT. > > What could the reason be audio in 1 direction is dropping? (Normally from > the Asterisk server to the mentioned SIP clients.) No clear information is > in the logs (it is like the call ended normally) and not all calls are > having problem (most not, but it happens to often for us to start using VoIP > more at the moment). > > To test if it was the firewall we disabled the firewall on the Asterisk > server and moved the Asterisk server before the other firewalls we have. > > What could the problem be? And even more important what could solve it > (and/or explain it)? > > Kind regards, > > Mark Scholten >
Hi Are the broadband connections to the target SIP extensions dedicated for VoIP or does any other traffic run over them? We tend to find that 80% of call quality issues are caused by the broadband connection. A good diagnostic tool to keep an eye on the broadband connections involved is smokeping http://oss.oetiker.ch/smokeping/ We find it an absolute godsend. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _____________________________________________________________________ -- 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
