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[shabbirabbasi92 at gma...]

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  ??