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<body class='hmmessage'><div dir='ltr'>Thanks for the information Anthony!<br><br><div><hr id="stopSpelling">Date: Thu, 6 Feb 2014 20:38:17 -0600<br>From: anthony.minessale@gmail.com<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] FS <-> SKYPE with G729. how to set annexb=no ?<br><br><p dir="ltr">The var is now begins rtp_ rather than sip_ this applies to all media related vars. </p>
<div class="ecxgmail_quote">On Feb 6, 2014 12:31 PM, "Yuriy Nasida" <<a href="mailto:nasida@live.ru">nasida@live.ru</a>> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">
<div><div dir="ltr">Hi there!<div><br></div><div>I know that it should be simple but when i add</div><div><pre style="font-family:monospace,Courier;padding:1em;border:1px dashed rgb(47,111,171);background-color:rgb(249,249,249);line-height:1.3em;font-size:12.727272033691406px;"><action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
</pre><div><br></div><div>nothing happens...</div><div><br></div><div>I didn't see fmtp:18 annexb=no in SDR of B-led</div><div><br></div><div><div>the callflow:</div><div><br></div><div>Skype --> FS ----G729a---> VOIP</div>
</div><div><br></div>Please advice<br>Thanks.</div> </div></div>
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