Hello all, I have a sip client that is register on one asterisk server, that asterisk server is routing the sip call to another asterisk server where it hops off to a pstn line via a X100P card. The call goes out but there is no audio on either side. I have checked the codecs on both servers to insure that that is not the issue, but I have been unable to find the problem. If someone might know the direction to look in i would appreciate a point in that direction. Thanks in advance for all assistance. Daniel Starks dlstarks@yahoo.com __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard antispam.yahoo.com/whatsnewfree