Hi, I've a FXO running Asterisk 1.8.32.1 registered to Freeswitch 1.4.15~1-1~wheezy+1. Majority of codecs (G722, ALAW, GSM, etc) work well, except for LIN16@16000h (SLIN16 in Asterisk). LIN16@8000h works great: v=0 o=root 739945760 739945761 IN IP4 192.168.1.2 s=Asterisk PBX 1.8.32.1 c=IN IP4 192.168.1.2 t=0 0 m=audio 18760 RTP/AVP 10 101 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[L16:70:8000:20:128000:1] 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [L16:70:8000:20:128000:1] ++++ is saved as a match 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[G722:9:8000:20:64000:1] 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[PCMA:8:8000:20:64000:1] But with LIN16@16000h codec can't be matched, even being on Freeswitch allowed list!: v=0 o=root 202962363 202962364 IN IP4 192.168.1.2 s=Asterisk PBX 1.8.32.1 c=IN IP4 192.168.1.2 t=0 0 m=audio 13330 RTP/AVP 118 3 0 8 101 a=rtpmap:118 L16/16000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:118:16000:20:0:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [GSM:3:8000:20:13200:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:70:16000:20:256000:1] With a ptime of 20ms, packet size is well below my MTU. Any hints?