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 ??
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