I'm currently running an office (about 25 phones) of Cisco 7960G's running SIP back to my asterisk box running 1.0.1. The asterisk box is attached to the Telco via a PRI (via a T100P). I'm getting complaints that the phone calls are cutting out on people (both parties) at random intervals, for anywhere from a half second to 5 seconds. In review the recordings, there is no cutout occurring on there, both parties are being recorded by asterisk. Any ideas? Paul