<div dir="ltr">Thanks.<div><br></div><div>Here is a pcap and the pastebin is at:</div><div><br></div><div><a href="https://pastebin.freeswitch.org/24299">https://pastebin.freeswitch.org/24299</a><br></div><div><br></div><div>I realized that if I wait about 10 seconds with the first call on hold, and don't even attempt to transfer, FS will hangup both legs of the original call. Also, there is an error before FS decides to hangup:</div><div><br></div><div>2015-07-08 15:54:36.617835 [CRIT] switch_core_io.c:173 sofia/internal/<a href="http://140@10.50.100.16:5060">140@10.50.100.16:5060</a> reading on a session with no media!<br></div><div><br></div><div>Appreciate your help.</div><div><br></div><div>Neven</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 8, 2015 at 11:03 AM, Giovanni Maruzzelli <span dir="ltr"><<a href="mailto:gmaruzz@gmail.com" target="_blank">gmaruzz@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">use 1.4.x latest (1.4.20), traces in pcap and pastebin<br><br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Wed, Jul 8, 2015 at 3:23 PM, Neven Boric <span dir="ltr"><<a href="mailto:nboric@yx.cl" target="_blank">nboric@yx.cl</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 class="h5"><div dir="ltr">Hi,<div><br></div><div>I have been using FS for a long time on a local server without issues, but now I want to move to a remote server to support some new usage scenarios. I'm trying to use inbound-bypass-media=true to keep the audio out of the server. This mostly works, but I have two different issues, depending on which FS version I use:</div><div><br></div><div>- I first tried with the 1.2 branch, as that was the version I was using locally. The issue with that version is that when a phone holds and then unholds a call, I get no audio on the phone that started the hold. I found option bypass-media-after-hold, but realized that it was not included in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from the master branch. Now FS correctly tries to restore direct media between the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in the last SDP to the phone that unholds, so I still get no audio. I haven't found a way to fix this.</div><div><br></div><div>- I also tried with the 1.4 branch and hold/unhold works correctly, but now attended transfer doesn't work, FS after some time ends the call on the side that was put on hold waiting to be transferred. This time seems to be inconsistent, and sometimes the transfer actually works, so maybe it is a timing issue. This seems similar/related to FS-4038.</div><div><br></div><div>I can get logs and SIP captures, what would be the preferred way to provide them (pcap, inline text, pastebin)? Or maybe it's better to go ahead and file a bug?</div><div><br></div><div>Best regards</div><span><font color="#888888"><div>Neven Boric</div></font></span></div>
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