Chris Maciejewski wrote: > Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - > audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - > 0x0 (nothing)
'us' does not include g726, so you have not configured your SIP user/peer to support G.726. > I note "Got unsupported a:fmtp in SDP offer" No, that is not relevant. Asterisk's SDP parser does not pay much attention to a:fmtp entries at this time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
