Further to my previous email, I have definitely established that the audio gets choppy only when the path includes sip and capi. PAP2 to Asterisk to MyNetFone to PSTN is fine. PAP2 to Asterisk MOH is fine. PBX (via capi) to Asterisk MOH is fine PBX (via capi) to Asterisk to PAP2 is choppy PBX (via capi) to Asterisk to MyNetFone to PSTN is choppy (PAP2 is a LinkSys FXS ATA) SIP phone, PBX, and MyNetFone are all configured to use alaw (G.711a), so transcoding should be almost irrelevant. Network bandwidth is not a problem because pure SIP calls are crystal clear - people I have called cannot tell the difference. I also have a X100 card which is providing timing. Nothing is sharing interrupts with anything. Any suggestions? Thanks James