<div dir="ltr">How are you getting it to work? Because it wasn't designed to work in that case.<div><br></div><div>
<p class=""><span class="">if</span><span class=""> (inbound_channel && inbound_session && !zstr(network_addr_a) && !zstr(network_addr_b) && !strcmp(network_addr_a, network_addr_b)), </span></p><p class=""><span class=""><br></span></p><p class=""><span class="">Guessing you luck out due to a proxy?</span></p></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 19, 2015 at 4:58 PM, jay binks <span dir="ltr"><<a href="mailto:jaybinks@gmail.com" target="_blank">jaybinks@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><div><div><div><div>Its also really good when you have <br><br></div>Box A ---- Box B ---- Box C <br><br></div>and Box B was used to setup the call, but is doing little more than proxying traffic.<br><br></div>Im using it in this way, where Box A is my carrier side SBC , Box C is my Customer side SBC <br>and I have call handling logic in Box B.<br><br></div>Once the call is established, Box B can get out of the path.<br></div>I do this so we can remove a potential point of failure.<br><br><br><br><div><div><br><br><div><br><br></div></div></div></div><div class="gmail_extra"><div><div class="h5"><br><div class="gmail_quote">On 20 February 2015 at 07:51, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</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">Simplify is only for dumping the entire call media and signaling off to prevent hair pinning of media. <div><br></div><div>Example:<div><br></div><div>BOX A -> BOX B -> BOX A, If this path were to take place a simplify operation could take place to send the media back to BOX A, This is handy if BOX A is on the east coast and BOX B is on the west coast. Cuts un-needed latency out of the path.</div></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Thu, Feb 19, 2015 at 9:24 AM, jay binks <span dir="ltr"><<a href="mailto:jaybinks@gmail.com" target="_blank">jaybinks@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><p dir="ltr">Can anyone give a working example of auto simplify ?? I can't make it work, but a media hook calling uuid_simplify works great.</p>
<p dir="ltr"> Bit odd.</p><div><div>
<div class="gmail_quote">On 20 Feb 2015 00:37, "jay binks" <<a href="mailto:jaybinks@gmail.com" target="_blank">jaybinks@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">hmmm seems I had to play further with <div><br></div><div>sip_route_uri<br></div><div>and </div><div>sip_invite_req_uri<br></div><div><br></div><div>to make this work, but it works now..</div><div><br></div><div><br></div><div>I do still wonder if simplify should use the request URI in the Refer-To header instead of what appears to be the SIP To URI.</div><div><br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 19 February 2015 at 23:12, jay binks <span dir="ltr"><<a href="mailto:jaybinks@gmail.com" target="_blank">jaybinks@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">Im bringing up a call ( "Freeswitch node A" ) where im using sip_route_uri to send the call via a specific node in my network "Freeswitch node B".<div><br></div><div>but I also want to get this "Freeswitch Node A" out of the call path after the call is answerd.</div><div>"simplify" works great for this.</div><div><br></div><div>however when I try and simplify a call, that was brought up with a sip_route_uri </div><div>my simplify fails ! :(</div><div><br></div><div>The Refer-To: sent by "FS node A" contains the SIP URI for my call , but has no reference to the details used in sip_route_uri meaning nothing was sent to "Freeswitch node B" </div><div><br></div><div>is this a bug, or am I missing something in the use case of these 2 features ??<span><font color="#888888"><br><div><div><br></div>-- <br><div>Sincerely<br><br>Jay</div>
</div></font></span></div></div>
</blockquote></div><br><br clear="all"><div><br></div>-- <br><div>Sincerely<br><br>Jay</div>
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</blockquote></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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