Hi Giovanni,<div><br></div><div>Our case 3 test was actually being bridged to a SIP endpoint (not in a conference, although</div><div>this probably should not matter). </div><div><br></div><div>We tested two type 3 cases:</div>
<div><br></div><div>1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP phone</div><div>2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN gateway</div><div><br></div><div>In both cases we are transiting FreeSWITCH, not terminating on it (could there be a</div>
<div>race condition when doing signaling coordination to remote SIP endpoints, not FreeSWITCH </div><div>itself?).</div><div><br></div><div>When we terminate Skype calls on FreeSWITCH this works (case 2). It also works</div>
<div>for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to Skype).</div><div><br></div><div>Also, we are running the very latest (well, as of this morning) git version, so that</div><div>may introduce one more variable (if you are running an older version).</div>
<div><br></div><div>Thanks for your help,</div><div><br></div><div>Jason<br><div><br><div class="gmail_quote">On Mon, Aug 23, 2010 at 6:36 PM, Giovanni Maruzzelli <span dir="ltr"><<a href="mailto:gmaruzz@celliax.org">gmaruzz@celliax.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Jason,<br>
<br>
with 3) - the problematic one - you mean an incoming call from a<br>
remote skype to the fs-skype, that is then bridged to an outbound sip<br>
call?<br>
<br>
I'm asking this because this is the most tested case, eg: in the<br>
gateway for the FS-conference, where the incoming skype calls are then<br>
bridged (through dialplan, like pressing "1" in the standard IVR) with<br>
the sip conference. And it works with no problems...<br>
<br>
Let me know more, please.<br>
<font color="#888888"><br>
-giovanni<br>
</font><div><div></div><div class="h5"><br>
On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords <<a href="mailto:jason@cloudtree.net">jason@cloudtree.net</a>> wrote:<br>
> Hi Giovanni,<br>
> We got some test cycles today and have found the following:<br>
> 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype<br>
> 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg is<br>
> FreeSwitch (internal extension)<br>
> 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP<br>
> In the last case:<br>
> 1) We hear Skype ringing until the B-leg (SIP) is answered - this is correct<br>
> 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the user<br>
> answering does not hang up, but the connection hangs up)<br>
> 3) The Skype client connection stays up with dead air until it is manually<br>
> hung up<br>
> We are running the latest code as of today.<br>
> Any insight you could provide will be most appreciated.<br>
> Thanks of all your work,<br>
> Jason<br>
> On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli <<a href="mailto:gmaruzz@celliax.org">gmaruzz@celliax.org</a>><br>
> wrote:<br>
>><br>
>> Jason,<br>
>><br>
>> please test with the latest git version.<br>
>><br>
>> Many messy changes to an already messy code, so maybe some bug or side<br>
>> effect (ok, bug) has been introduced.<br>
>><br>
>> commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090<br>
>> Author: Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com">gmaruzz@gmail.com</a>><br>
>> Date: Fri Aug 13 16:19:20 2010 -0500<br>
>><br>
>> skypopen: now answer a call only when directed to do it (before<br>
>> was trying to answer any incoming call). Lot of changes to a messy<br>
>> part, so maybe some problem will come out...<br>
>><br>
>> Signed-off-by: Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com">gmaruzz@gmail.com</a>><br>
>><br>
>> -giovanni<br>
>> --<br>
>> Sincerely,<br>
>><br>
>> Giovanni Maruzzelli<br>
>> Cell : +39-347-2665618<br>
>><br>
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><br>
><br>
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><br>
><br>
<br>
<br>
<br>
</div></div>--<br>
<div><div></div><div class="h5">Sincerely,<br>
<br>
Giovanni Maruzzelli<br>
Cell : +39-347-2665618<br>
<br>
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