Hi, Is there any information that can be gathered from the logs about why a SIP call was dropped/terminated without either side hanging up? I've run asterisk pretty verbose and I guess I haven't seen anything that pops out at me yet. I'm trying to diagnose why some clients are getting dropped calls every so often.
Thanks. -- James _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
