<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">What do you expect to be hearing when eavesdropping on a parked channel?<div><br><div><div>On Mar 21, 2014, at 9:05 PM, Yan Brenman <<a href="mailto:ybrenman@xopnetworks.com">ybrenman@xopnetworks.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div bgcolor="#FFFFFF" text="#000000" style="font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><font size="-1">I just tried the "master" branch and situation there with the "eavesdrop" application is absolutely the same as it is on the "1.4.beta" branch.<br>It doesn't work for the WebRTC clients and it does work for the non-WebRTC clients (I used jitsi client to test). Having said that I was only<br>getting audio working with "eavesdrop" for non-WebRTC clients but I am pretty sure if I merge in some code I used to from the video-media-bug<br>branch - I will get the video too.<span class="Apple-converted-space"> </span><br><br>Just to describe you my test scenario - I am placing and inbound leg with I am "answering" and "park". Then from the FS_CLI I am "originating" an<br>outbound leg which is "eavesdrop" on the inbound "parked" leg. I hope you don't see anything wrong with it.<br><br>Thanks again<br><br></font><div class="moz-cite-prefix">On 3/21/2014 8:38 AM, Ken Rice wrote:<br></div><blockquote cite="mid:CF51B923.7AD47%25krice@freeswitch.org" type="cite"><font face="Monaco, Courier New"><span style="font-size: 11pt;">You should be testing webrtc stuff on the master branch... Theres been a lot of changes going on in tree.. The video-media-bug branch is probably incomplete... I don’t think its even close to ready for merging at this point...<br><br><br>On 3/18/14 2:01 PM, "Yan Brenman" <<a moz-do-not-send="true" href="x-msg://79/yb91423@gmail.com">yb91423@gmail.com</a>> wrote:<br><br></span></font><blockquote><font face="Monaco, Courier New"><span style="font-size: 11pt;">Hello,<br><br>I was trying to get the eavesdrop application work on the 1.4.beta branch for the WebRTC clients (works great for all other<br>types of clients) and could not. Neither audio nor video (I merged the code from the video-media-bug branch for the video)<br>is going through. For the client I tried both - sipml5 and jssip clients.<br><br>Please let me know if there is any other piece of information I can provide.<br><br>I will be more then happy to do any kind of code changes and testing - just need some starting points. Especially considering<br>that I am very new to the WebRTC.<br><br>Every piece of advice is greatly appreciated. <br><br>Thanks<br>Yan<br><br></span></font></blockquote></blockquote></div></blockquote></div><br></div></body></html>