<div dir="ltr">Hello,<br><br>First of all sorry for my previous e-mauil because it was incomplete<br>Here you have the complete one<br><br>I am using freeswitch as class IV switch.<br><br>I have observed that updates from SIP servers are nos passed upstream in the following scenario.<br><br>IVR---(profile1)---FS---(profile2)----SBC----------GW1-----PSTN<br> |<br> |----------GW2-----PSTN<br><br>1. A call arrive from GW1 to the IVR<br>2. The IVR calls to a second phone number reachable by means of GW2<br>3. Once the IVR has the two calls are established the IVR do SIP bridging in order the RTP doesn't pass throuhgt it <br>and then send two updates to FS (one for each call) to switch the calls together.<br>4.FS undertands the updates retaining the RTP and switching both calls, I mean, the user who made the first call is now in comunication with the user behind GW2.<br><br>The question is:<br> <br>Why FS retain the RTP instead of send the UPDATES towards the SBC in order this one be which retain the RTP?<br><br>Maybe is because the IVR and the SBC are not in the same profile?<br><br><br>Thanks if someone has faced with this matter and have any explanation<br><br>Regards<br><br></div><div class="gmail_extra"><br><div class="gmail_quote">2017-03-15 13:15 GMT+01:00 Jose Serrano <span dir="ltr"><<a href="mailto:jjserranor@gmail.com" target="_blank">jjserranor@gmail.com</a>></span>:<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>Hello,<br><br></div>I am using freeswitch as class IV switch.<br><br></div>I have observed that updates from SIP servers are nos passed upstream in an scenario as the following<br><br></div>IVR------FS-------SBC---------<wbr>-GW1<br></div> |<br>_____GW" <br></div>
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