I think I might be experiencing a similar problem here... A friend of mine dialed into one of my conference rooms with only the GSM codec enabled on his mobile SIP client (Linphone). Despite the fact that I offered GSM as one of the codecs, FS only appeared to offer G711u and the call failed. When he set G711u as an option on his end, he was able to connect. Late negotiation is enabled. Some relevant settings: internal.xml: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="inbound-codec-negotiation" value="greedy"/> vars.xml: <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> SIP transaction info follows: ------------------------------------------------------------------------ recv 765 bytes from udp/[68.47.xx.xx]:5060 at 06:59:23.381436: ------------------------------------------------------------------------ INVITE sip:3030@myswitch.org:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060;rport;branch=z9hG4bK1252688933 From: "resist0r" <sip:resist0r@68.47.xx.xx>;tag=539248952 To: <sip:3030@myswitch.org:5080> Call-ID: 1476867094 CSeq: 20 INVITE Contact: <sip:resist0r@68.47.xx.xx> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.4.3 (eXosip2/3.1.0) Subject: Phone call Expires: 120 Content-Length: 234 v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/22050 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ------------------------------------------------------------------------ send 354 bytes to udp/[68.47.xx.xx]:5060 at 06:59:23.381660: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.15:5060;rport=5060;branch=z9hG4bK1252688933;received=68.47.xx.xx From: "resist0r" <sip:resist0r@68.47.xx.xx>;tag=539248952 To: <sip:3030@myswitch.org:5080> Call-ID: 1476867094 CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f2cf68b 2011-11-20 18-40-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-25 01:59:23.365421 [NOTICE] switch_channel.c:920 New Channel sofia/external/resist0r@68.47.xx.xx [b9d2afd7-e0ba-4168-acee-29ccd2dc4791] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5356 Channel sofia/external/resist0r@68.47.xx.xx entering state [received][100] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5367 Remote SDP: v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMA:8:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[GSM:3:8000:20:13200] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[iLBC:97:8000:30:13330] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:2869 Set Codec sofia/external/resist0r@68.47.xx.xx PCMU/8000 20 ms 160 samples 64000 bits Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com