I wonder if anyone else is having these problems. We are running Asterisk 1.2.17, with an assortment of SIP users and peers. This is running on an 600 MHz P3 with CentOS 4.4, and worked properly in Asterisk 1.2.15. Nothing else running on the server except the usual support stuff like sshd, a mostly idle httpd, and no GUI. app_dictate works fine for recording, but on some calls during playback the audio jumps around, playing fragments of the file. Using the fast playback mode sometimes works, sometimes causes the jumping around to get worse. Incoming calls to the Dictate() application from different SIP carriers and different hard and soft phones give drastically different results. For instance, dialing in via an 01 Communications DID (resold by Broadvoice) at 831-713-4569 fails on playback (as described, just fragments of audio) every time. Dialing in via a Broadwing DID (resold by Vitelity) at 831-621-1913 works. Calling from a Grandstream phone fails, from a Cisco 7960 works most of the time, from a Motorola VT-1005 ATA always works. All other playback modes including MOH work fine. I have some clue, but not enough. Any ideas?