Re: [Freeswitch-users] Try to get rid of Speex codec on invite

You are making the wrong conclusion. The SDP is being verbose. That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged. See the codec list in the m= line? /b On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:

Hello,

when our Freeswitch sends an INVITE to a phone, it only offers speex codec for voice: m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16