Hello list. I have the next problem with my Freeswitch. I’m using a Sangoma D100 transcoding card, but ‘im having problems with G729. This is my scenario. GatewayA ---------> FreeSwitch+D100 -----------> GatewayB (10.0.0.220) (10.0.0.148) (10.0.0.222) Gateway A is calling through Freeswitch to Gateway B. The SDP of the initial INVITE message has the parameters : “a=rtpmap:18 G729/8000” “a=fmtp:18 annexb=no” Then the INVITE transmitted to the gateway B it also has the same parameters. The problem is with the “200 - OK” message that goes from FreeSwitch to Gateway A. The “200 - OK” coming from Gateway B has the parameter “a=fmtp:18 annexb=no” in the SDP, but Freeswitch is not attaching this parameter to the “Leg A” and this is causing a one-way audio in the call. The weird thing is when i unload the “mod_sangoma_codec” and load the “mod_g729”, this time the “200 OK” message from the Freeswitch is using the “a=fmtp:18 annexb=no” to the Leg A, and the call is successfully established without one way audio problem. I really don’t know what could be happening, I ask the Sangoma support but thay said that this is a Freeswitch bug. Can someone help me here? These are part of my configuration files. “default.xml” <extension name="to_prueba2"> <condition field="network_addr" expression="^10\.0\.0\.220$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="nolocal:absolute_codec_string=G729,G723"/> <action application="set" data="sip_invite_domain=10.0.0.222"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:18 G729/8000,a=fmtp:18 annexb=no"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}@10.0.0.222"/> <action application="answer"/> </condition> </extension> <extension name="to_prueba1"> <condition field="network_addr" expression="^10\.0\.0\.222$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=10.0.0.220"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}@10.0.0.220"/> <action application="answer"/> </condition> </extension> Part of the “interior.xml” file: <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <param name="inbound-late-negotiation" value="true"/> <!--set to 'greedy' if you want your codec list to take precedence --> <param name="inbound-codec-negotiation" value="generous"/> Y finalmente en los codec preferentes tengo (vars.xml) <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G723,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,G723"/> Thanks in advance. Regards, Ricardo.-