Hello, I am using * as a SIP proxy with several SIP clients. The SIP clients are SJPhone Soft phones. All clients are inside a firewall and the Server is inside too. All is working fine, but the speech sounds like Micky Mouse. If you feed one client?s (Mic) input with a permanent tone i.e. a 440 Hz Sinus wave it?s frequency on the (Speaker) output of the client you are connected to is shifted to a higher frequency. In addition to this you can hear drop outs. Obviously the samling rate on the sender?s side does not fit the receivers rate. I do not understand this because both phones are using G.711 ALAW. Taking a look at *?s channels with the help of it?s command line interface shows ALAW for both channels too. I set reinvite=no in the sip.conf file, because SJPhone did not support this and the connection broke down. So if I understand things right the conversion error could also be caused by *, because it stays inside the rtp connection. Does anybody know something about this phenomena?? Thanks in advance joerg. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041111/6776d5aa/attachment.htm