<div dir="ltr"><div><span style="font-size:12.8px">I got it from the following URL.</span></div><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px"><a href="https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation">https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation</a></span><br></div><span style="font-size:12.8px"><div><span style="font-size:12.8px"><br></span></div>---------- Forwarded message ----------</span><br style="font-size:12.8px"><span style="font-size:12.8px">From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span><br style="font-size:12.8px"><span style="font-size:12.8px">To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>></span><br style="font-size:12.8px"><span style="font-size:12.8px">Cc: </span><br style="font-size:12.8px"><span style="font-size:12.8px">Date: Tue, 17 May 2016 16:45:16 -0400</span><br style="font-size:12.8px"><span style="font-size:12.8px">Subject: Re: [Freeswitch-users] Disable annexb for g729</span><br style="font-size:12.8px"><div style="font-size:12.8px;word-wrap:break-word">It was spread about pretty widely when we went to 1.4 but thats a couple years ago now. It should be documented on the wiki but may have fallen off during the transition to confluence as 1.2 is quite old. Where did you see docs for the sip_ var?</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, May 17, 2016 at 3:51 PM, 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 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" target="_blank">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="HOEnZb"><div class="h5"><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><div><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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