[Freeswitch-users] mod_skypopen is awesome issue with demo-ivr conference

Giovanni Maruzzelli gmaruzz at celliax.org
Sun Jun 5 18:38:31 MSD 2011 

:)
again, read the wiki page.
Messages use the CHAT standard freeswitch interface, same used by
dingaling ans sofia.
So, you can simply make an application that routes messages between
skype, google and sip, eg, using the ESL.
-giovanni

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

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

More information about the FreeSWITCH-users mailing list