<font color="#333333"><font size="2"><font face="verdana,sans-serif">Sorry, I didn't make it clear. I did search google and found "general" information of skype being behind NAT or firewall. But I was looking for something that is capable of turning that feature off. So I tossed the questions here hoping people having done integration with Skype would know better. <br>
<br>I also thought of some feature that I am not sure if I should posted here or open a new thread. I was thinking of integrating the mod_skypopen with mod_digaling or some other sort so that when the skype client receive messages, we can transfer it to a designated XMPP client with a 2 way text design in mind.<br>
<br><br>Thanks,<br><br>Henry<br></font></font></font><br><div class="gmail_quote">On Sun, Jun 5, 2011 at 2:21 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">If you're behind a nat or a firewall you can't be supernode.<br>
If you're directly on the internet (eg: you have a real routable IP<br>
number) on the machine, and you are not behind a firewall (eg: you<br>
have all incoming ports open) then you *may* become a supernode.<br>
Btw: google is your friend, those are informations easily available<br>
with a search for "skype supernode".<br>
<font color="#888888">-giovanni<br>
</font><div><div></div><div class="h5"><br>
On 6/5/11, Henry Huang <<a href="mailto:b_ball_henry@hotmail.com">b_ball_henry@hotmail.com</a>> wrote:<br>
> One thing I didn't see it covered in the wiki. Is the skype client we use<br>
> for the mod_skypopen having the risk of being used as a "supernode"? If so,<br>
> is there ways to disable it?<br>
><br>
> Thanks,<br>
><br>
> Henry<br>
><br>
> On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli<br>
> <<a href="mailto:gmaruzz@gmail.com">gmaruzz@gmail.com</a>>wrote:<br>
><br>
>> You need to set the variable<br>
>> 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.<br>
>><br>
>> This is my<br>
>> /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :<br>
>><br>
>> <include><br>
>> <extension name="conf_via_skype"><br>
>> <condition field="destination_number" expression="^5004$"><br>
>> <action application="set"<br>
>> data="skype_add_outband_dtmf_also_when_bridged=true"/><br>
>> <action application="set" data="ringback=$${us-ring}"/><br>
>> <action application="sleep" data="500"/><br>
>> <action application="bridge"<br>
>> data="sofia/${use_profile}/<a href="mailto:888@conference.freeswitch.org">888@conference.freeswitch.org</a>"/><br>
>> </condition><br>
>> </extension><br>
>> </include><br>
>><br>
>> You can copy and paste it in the same file.<br>
>><br>
>> You then will edit the file<br>
>> /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the<br>
>> line:<br>
>><br>
>> <entry action="menu-exec-app" digits="1" param="bridge<br>
>> sofia/$${domain}/<a href="mailto:888@conference.freeswitch.org">888@conference.freeswitch.org</a>"/><br>
>><br>
>> with the line:<br>
>><br>
>> <entry action="menu-exec-app" digits="1" param="transfer 5004 XML<br>
>> default"/><br>
>><br>
>> and it will work as you expect, both with inbound skype and with inbound<br>
>> sip.<br>
>><br>
>> -giovanni<br>
>><br>
>><br>
>> On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com">gmaruzz@gmail.com</a>><br>
>> wrote:<br>
>> > As per the wiki page, and as is clearly written in the console and the<br>
>> > logfile, if an incomong skype call is bridged to an outbound call, the<br>
>> > dtmf is by default passed only in band (eg: as audio).<br>
>> > If you want to have it working with the fs developer conference,<br>
>> > create an extension that goes there and add to it the variable that's<br>
>> > in the wiki page (can't remember the name, is something like<br>
>> > "dtmf-outband-also-when-bridged=true").<br>
>> > Please check the wiki page for the variable name.<br>
>> > Reason for this is that the in band dtmf cannot be shut up, and often<br>
>> > is detected by the remote party. But in this particular case of the fs<br>
>> > developer conference, the remote party is setup to not detect inband<br>
>> > in sip (or anyway, it does not detect it).<br>
>> > Long story short: create an extension that sets that variable and goes<br>
>> > to the fs conference and you'll be all set.<br>
>> > Tomorrow I'll post here an example from my dialplan.<br>
>> > -giovanni<br>
>> ><br>
>> > On 6/3/11, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<br>
>> >> Try transferring to a local conference on your machine and see what<br>
>> happens.<br>
>> >> Watch the fs_cli and see if the DTMFs show up or not. From there we can<br>
>> see<br>
>> >> what's up...<br>
>> >><br>
>> >> -MC<br>
>> >><br>
>> >> On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang<br>
>> >> <<a href="mailto:b_ball_henry@hotmail.com">b_ball_henry@hotmail.com</a>>wrote:<br>
>> >><br>
>> >>> But the DTMF press worked very well while I was under the Demo IVR,<br>
>> even<br>
>> >>> when I press "1001" I was able to get connected to extension 1001. It<br>
>> >>> didn't<br>
>> >>> work only after I was transferred to the conference selection.<br>
>> >>><br>
>> >>> Henry<br>
>> >>><br>
>> >>> On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins<br>
>> >>> <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>>wrote:<br>
>> >>><br>
>> >>>> The monkeys have stopped screaming for everyone. :( I have been<br>
>> looking<br>
>> >>>> for an alternative sound file for this but haven't found anything I<br>
>> >>>> really<br>
>> >>>> like.<br>
>> >>>><br>
>> >>>> Not sure about the DTMF, but from what I understand there have been<br>
>> >>>> issues<br>
>> >>>> with sending DTMFs from the Skype client. I haven't tried it myself<br>
>> >>>> so<br>
>> I<br>
>> >>>> will defer to those who have.<br>
>> >>>><br>
>> >>>> -MC<br>
>> >>>><br>
>> >>>> On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang<br>
>> >>>> <<a href="mailto:b_ball_henry@hotmail.com">b_ball_henry@hotmail.com</a>>wrote:<br>
>> >>>><br>
>> >>>>> Hi:<br>
>> >>>>><br>
>> >>>>> The new interactive installation for mod_skypopen is a piece of<br>
>> >>>>> cake.<br>
>> >>>>> Made live much easier. I was so excited and tried it out right after<br>
>> the<br>
>> >>>>> installation was done. But I found a few issues calling the demo-ivr<br>
>> >>>>> from<br>
>> >>>>> skyTpe client.<br>
>> >>>>><br>
>> >>>>> 1. The screaming monkey does not scream anymore.<br>
>> >>>>> 2. The conference that connects to the <a href="http://freeswitch.org" target="_blank">freeswitch.org</a> doesn't<br>
>> respond to<br>
>> >>>>> dtmf pressed on the skype client. Ever since the second client is<br>
>> >>>>> connected<br>
>> >>>>> to the conference, the conference tell the client it's muted. So I<br>
>> tried<br>
>> >>>>> push 0 and any other key to see if I get to unmute. But no, non of<br>
>> the<br>
>> >>>>> key<br>
>> >>>>> press works.<br>
>> >>>>><br>
>> >>>>> If anyone have previous experiences on these and fixed them, please<br>
>> do<br>
>> >>>>> share your methods.<br>
>> >>>>><br>
>> >>>>> Sound quality with the new virtual sound drive is very good.<br>
>> >>>>><br>
>> >>>>> Thanks,<br>
>> >>>>><br>
>> >>>>> Henry Huang<br>
>> >>>>> US: +1(818)6885508 | 台灣(Taiwan): +886 933847619<br>
>> >>>>> Contact Me [image:<br>
>> >>>>> LinkedIn]<<a href="http://www.linkedin.com/profile/view?id=4590380" target="_blank">http://www.linkedin.com/profile/view?id=4590380</a>> [image:<br>
>> >>>>> Facebook] <<a href="http://www.facebook.com/profile.php?id=595148342" target="_blank">http://www.facebook.com/profile.php?id=595148342</a>> [image:<br>
>> >>>>> Twitter] <<a href="http://twitter.com/unicsolution" target="_blank">http://twitter.com/unicsolution</a>><br>
>> >>>>> IM [image: Google Talk] <a href="mailto:red_rain_seven@gmail.com">red_rain_seven@gmail.com</a> [image:<br>
>> >>>>> Skype]unicsolution [image:<br>
>> >>>>> MSN] <a href="mailto:b_ball_henry@hotmail.com">b_ball_henry@hotmail.com</a><br>
>> >>>>><br>
>> >>>>><br>
>> >>>>> _______________________________________________<br>
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>> >>>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>> >>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >>>>><br>
>> >>>>><br>
>> >>>><br>
>> >>>> _______________________________________________<br>
>> >>>> FreeSWITCH-users mailing list<br>
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>> >>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >>>><br>
>> >>>><br>
>> >>><br>
>> >>> _______________________________________________<br>
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>> >>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >>><br>
>> >>><br>
>> >><br>
>> ><br>
>> > --<br>
>> > Sent from my mobile device<br>
>> ><br>
>> > Sincerely,<br>
>> ><br>
>> > Giovanni Maruzzelli<br>
>> > Cell : +39-347-2665618<br>
>> ><br>
>><br>
>><br>
>><br>
>> --<br>
>> Sincerely,<br>
>><br>
>> Giovanni Maruzzelli<br>
>> Cell : +39-347-2665618<br>
>><br>
>> _______________________________________________<br>
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>><br>
><br>
<br>
</div></div>--<br>
<div class="im">Sent from my mobile device<br>
<br>
Sincerely,<br>
<br>
Giovanni Maruzzelli<br>
Cell : +39-347-2665618<br>
<br>
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