Hi,<div><br></div><div>All these solutions only work for the specific case that the re-INVITE is done to match an outgoing call (bridge / transfer). What I&#39;m looking for is a way to initiate a re-negotiation, not necessarily because the call is going to be bridged. For example to change the codec prior to entering a conference.</div>
<div><br></div><div>Greetings,</div><div>Roland<br><br><div class="gmail_quote">2012/1/23 Zohair Raza <span dir="ltr">&lt;<a href="mailto:engineerzuhairraza@gmail.com">engineerzuhairraza@gmail.com</a>&gt;</span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div>Hi, </div><div><br></div><div>Try setting bypass_media_after_bridge=true</div><div><br></div><div>with <span>&lt;param name=&quot;inbound-codec-</span><span>negotiation&quot; value=&quot;greedy&quot;/&gt; </span>in sofia profile and then pass absolute_codec_string </div>


<div>before bridge on the new leg like </div><div><br></div><div>&lt;action application=&quot;export&quot; data=&quot;nolocal:absolute_codec_string=PCMA,PCMU&quot;/&gt;</div>
<div><br></div><div>bypass_media_after_bridge=true is a bit similar to canreinvite of asterisk</div><div><br></div><div><div dir="ltr">Regards,<span class="HOEnZb"><font color="#888888"><br>Zohair Raza<div><br></div><div>
<br></div></font></span></div><div><div class="h5"><div class="gmail_quote">On Mon, Jan 23, 2012 at 1:59 PM, Muhammad Shahzad <span dir="ltr">&lt;<a href="mailto:shaheryarkh@googlemail.com" target="_blank">shaheryarkh@googlemail.com</a>&gt;</span> wrote:<br>



<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Not sure but i think FS can&#39;t initiate re-invite, however if caller (or callee) initiate a re-invite then FS responds accordingly, including codec renegotiation. <div>



<br></div><div>Thank you.</div><div><br><br><div class="gmail_quote">
2012/1/23 Roland Hänel <span dir="ltr">&lt;<a href="mailto:roland@haenel.me" target="_blank">roland@haenel.me</a>&gt;</span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">



<div><div>Hi,<div>
<br></div><div>Is there a way to trigger a SIP re-INVITE for a channel that has already been answered, and to make FreeSwitch offer a certain list of codecs to the endpoint?</div><div><br></div><div>Let&#39;s assume the call is answered with PCMA/PCMU, then the user enters an IVR, now we know that he&#39;s going to join a conference. For the conference, I might want to enable G.722 and H.264 video (which should not generally be enabled for all incoming calls). I found stuff about codec-related channel variables, but no &#39;trigger&#39; to tell FS to make a codec re-negotiation.</div>





<div><div><br></div>Greetings,<div>Roland</div><div><br></div><br>
</div>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br>Muhammad Shahzad<br>-----------------------------------<br>CISCO Rich Media Communication Specialist (CRMCS)<br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Gruß,<div>Roland</div><div><br></div><br>
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