i am trying to receive a call from freeswitch without transcoding , asterisk and freeswitch are installed on same machine

in asterisk cli with sip set debug on

v=0
o=FreeSWITCH 1442495774 1442495775 IN IP4 127.0.0.1
s=FreeSWITCH
c=IN IP4 127.0.0.1
t=0 0
m=audio 28840 RTP/AVP 98 13
a=rtpmap:98 L16/16000
a=ptime:20
Found RTP audio format 98
Found RTP audio format 13
Found audio description format L16 for ID 98
chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer!

is it possible to receive this call and pass it to chan_dongle ??