I use g723.1 for my SIP to SIP calls. To place calls to * i do a SIP_CODEC=ulaw. I have two clients, and ATA (v2.16) and MSN. The problem I have is with the ATA. When I dial an * app (voicemail for example) from the ATA , I keep getting the following NOTICE: rtp.c (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible. The phone call drops after a few seconds. I don?t get the problem with MSN. Does anyone know what configuration do I need to change on the ATA. I have looked at the rfc3389 and the ATA admin guide and couldn?t figure it out. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/83edc245/attachment.htm