> I've written about this issue several times, but have not yet found any > solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones > are primarily Snom 300's but I also have a couple of headset phones > connected to Grandstream HT286 SIP adapters. I have 8 offices, each has > it's own asterisk server all running the same versions of asterisk and > Zaptel. Only difference is that one office uses a Digium TDM 8-port > card and the other branches use 4-port Rhino cards with only 2 ports in > use. What happens is that periodically we will be in a call and the > call will just drop. It's usually within the first couple of minutes of > the call. The calls can be either incoming or outgoing. The phenomenon > affects both the Snoms and the Grandstreams. Along with the dropped > call issue, we periodically have a problem where a person we call or a > person that calls in cannot hear the person in the our office, but the > person in our office can hear the remote person fine. > > All of the phones are on the same physical network as the asterisk > server. There is no NAT, no Firewall, VLAN, etc. between the phones and > the server. I have tried running sip debugs on the calls, but on the > off chance that my logs catch either a drop or a one-way audio, the sip > debug looks like just a normal call. > > Is there any setting that might cause both one-way audio and dropped calls? > > Thanks, > Brent Davidson
Join the club. I've experienced the same with various strains on 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). This issue is truly random and debugging reveals nothing. I run an all SIP environment with same results. My solution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses -- _____________________________________________________________________ -- 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
