We have a carrier the mandates that annex b be disabled on the invites to them. I have added the following to the default dialplan but nothing seems to change the invite on the B leg to the carrier. <extension name="disable-annexB" continue="true"> <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)(.*)( 18 )(.*)/s"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> </condition> </extension> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> and <action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18 annexb=no,absolute_codec_string=^^:G729:PCMU:PCMA}sofia/gateway/carrierGW/$1"/> inbound-late-negotiation is set to true. The logs show that it is exporting: [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] to event The SDP does not reflect: Local SDP: v=0 o=FreeSWITCH IN IP4 10.10.X.X s=FreeSWITCH c=IN IP4 10.10.X.X t=0 0 m=audio 20022 RTP/AVP 18 0 8 101 13 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Any help would be greatly appreciated. Thanks!