<div dir="ltr"><div><span style="font-size:12.8px">Thanks Michael! That did the trick. Did I miss that somewhere?</span></div><span style="font-size:12.8px"><div><span style="font-size:12.8px"><br></span></div>From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span><br style="font-size:12.8px"><br style="font-size:12.8px"><div style="font-size:12.8px;word-wrap:break-word"><div><span style="color:rgb(51,51,51);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14px"><action application="export" data="rtp_append_audio_sdp=a=fmtp:18 annexb=no"/></span></div><div><span style="color:rgb(51,51,51);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14px"><br></span></div><div><font color="#333333" face="Helvetica Neue, Helvetica, Arial, sans-serif"><span style="font-size:14px">between 1.2 and 1.4 the var got renamed from sip_ to rtp_</span></font></div><div><font color="#333333" face="Helvetica Neue, Helvetica, Arial, sans-serif"><span style="font-size:14px"><br></span></font></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, May 17, 2016 at 10:57 AM, Mike Rice <span dir="ltr"><<a href="mailto:mrice0118@gmail.com" target="_blank">mrice0118@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5"><div dir="ltr"><div>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. </div><div><br></div><div><extension name="disable-annexB" continue="true"></div><div> <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)(.*)( 18 )(.*)/s"></div><div> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/></div><div> </condition></div><div></extension></div><div><br></div><div><action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/></div><div><br></div><div>and</div><div><br></div><div><action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18</div><div>annexb=no,absolute_codec_string=^^:G729:PCMU:PCMA}sofia/gateway/carrierGW/$1"/></div><div><br></div><div><br></div><div>inbound-late-negotiation is set to true. The logs show that it is exporting:</div><div><br></div><div>[DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable]</div><div>EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX)</div><div>[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</div><div>[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</div><div><br></div><div>The SDP does not reflect:</div><div><br></div><div>Local SDP:</div><div>v=0</div><div>o=FreeSWITCH IN IP4 10.10.X.X</div><div>s=FreeSWITCH</div><div>c=IN IP4 10.10.X.X</div><div>t=0 0</div><div>m=audio 20022 RTP/AVP 18 0 8 101 13</div><div>a=rtpmap:18 G729/8000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div><div><br></div><div>Any help would be greatly appreciated. Thanks!</div></div>
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