Hello, I have a remote sip phone (Linksys SPA3102) which only supports PCMU. When I call to this remote sip phone i get a 406 error that opus is not supported as shown by the sip trace below. However, if I force the codec to absolute like this {absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 the call works fine. Is there anyway I can make FreeSWITCH to automatically transcode without forcing the codec string in the dial plan? The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and outbound_codec_prefs=PCMU,PCMA,GSM ---------------------------siptrace-------------------------------- recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: ------------------------------------------------------------------------ SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q CSeq: 87372504 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: ------------------------------------------------------------------------ ACK sip:303@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej Max-Forwards: 68 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 CSeq: 87372504 ACK Content-Length: 0 ------------------------------------------------------------------------ 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel sofia/internal/303@192.168.1.5 entering state [terminated][406] 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Thanks Rajil