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!