[Freeswitch-users] Conference Questions

Fri Jan 15 13:07:33 PST 2010

Great!

Thanks Anthony, really appreciate the help.

Alfredo Q-V

On Fri, Jan 15, 2010 at 3:28 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> look up the originate api instead of conference dial
>
> originate /dial/string conference:myconf+flags{foo} inline
>
>
> On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> Anthony:
>>
>> I think I spoke too soon. I looked at the example; but I was under the
>> impression based on your previous comment that I would be able to
>> invoke the conference dial api command over HTTP and specify the flags
>> as part of the dial string. The wiki page has the following for the
>> API call I am thinking:
>>
>> dial
>>
>> Dial a destination via a specific endpoint (ie. call mom from the
>> conference).
>>
>> Usage: conference <confname> dial [{dial string
>> options}]<endpoint_module_name>/<destination> [<callerid_number>
>> [<callerid_name>]]
>>
>> I would like to specify the privilege in this api call. Is that
>> doable? If not how could I accomplish it? Would I be able to pass the
>> flags in {dial string options}?
>>
>> Thanks in advance for the help,
>>
>> Alfredo
>>
>> On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>> > Awesome! That's even nicer.
>> >
>> > Appreciate it.
>> >
>> > Alfredo Q-V
>> >
>> > On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >> the flags are set as part of the dial string so you can easily choose
>> >> that,
>> >> int the example I told you to look at notice the +flags{} bit at the
>> >> end of
>> >> some of the dial strings.
>> >>
>> >>
>> >> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil
>> >> <lawwton at gmail.com> wrote:
>> >>>
>> >>> Appreciate the fast response Anthony.
>> >>>
>> >>> Response or ideas on how to implement a) ?
>> >>>
>> >>> a) It seemed to me like the way to setup the moderator of the
>> >>> conference is by setting a parameter in the DialPlan and specifying
>> >>> based on a condition who the moderator is, say for instance the
>> >>> destination number. That's fine and it makes sense, however, say that
>> >>> I am creating a new conference and I want to have 3 participants where
>> >>> one of them is the moderator. What would I have to do to specify that
>> >>> person A dialing for example number xxx-xxx-xxxx is the moderator (via
>> >>> HTTP)? Would I have to create my own call to the system and add say an
>> >>> entry to DialPlan with the right parameter for the moderator, then
>> >>> create the conference?
>> >>>
>> >>> Thanks in advance,
>> >>>
>> >>> Alfredo Q-V
>> >>>
>> >>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > look at the "mad boss" extension in the default dialplan
>> >>> > conf/dialplan/default.xml to see how to craft an all-hands
>> >>> > conference.
>> >>> > otherwise individual calls to originate to send people to the
>> >>> > conference
>> >>> > is
>> >>> > also ok.
>> >>> >
>> >>> >
>> >>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil
>> >>> > <lawwton at gmail.com> wrote:
>> >>> >>
>> >>> >> Hello:
>> >>> >>
>> >>> >> I've been using asterisk for a little bit over three years now. A
>> >>> >> couple of months ago I found out about freeswitch, took a look at
>> >>> >> it,
>> >>> >> thought it was interesting and moved on. A few weeks ago, I started
>> >>> >> looking at a project I've been wanting to work on for quite a while
>> >>> >> using conferences and started exploring systems and different
>> >>> >> approaches. Based on the requirements I have, I decided to use
>> >>> >> freeswitch. It seemed like it had the best support for conferencing
>> >>> >> so
>> >>> >> I went for it. According to some documentation I found it also
>> >>> >> seems
>> >>> >> to allow for more concurrent calls than asterisk which is an added
>> >>> >> bonus.
>> >>> >>
>> >>> >> I got a server ready, installed FC8 on it which is what I have in
>> >>> >> production now, unpacked freeswitch there and so far it's running
>> >>> >> beautifully. Very painless process really to get it installed, I
>> >>> >> was
>> >>> >> happy to see that. Configuration seems a bit different since it's
>> >>> >> XML;
>> >>> >> but being a developer myself I can see many advantages to having
>> >>> >> done
>> >>> >> that in the future as the system scales and grows in complexity.
>> >>> >>
>> >>> >> Sorry for the long introduction, getting to my question now. So ...
>> >>> >> What I want to be able to do is the following:
>> >>> >>
>> >>> >> Create and control conferences via the HTTP API. I've been reading
>> >>> >> a
>> >>> >> bit for the past two days the documentation and I am becoming more
>> >>> >> familiar now with how things are done using ESL, the support for
>> >>> >> PHP,
>> >>> >> perl and I believe others.
>> >>> >>
>> >>> >> a) It seemed to me like the way to setup the moderator of the
>> >>> >> conference is by setting a parameter in the DialPlan and specifying
>> >>> >> based on a condition who the moderator is, say for instance the
>> >>> >> destination number. That's fine and it makes sense, however, say
>> >>> >> that
>> >>> >> I am creating a new conference and I want to have 3 participants
>> >>> >> where
>> >>> >> one of them is the moderator. What would I have to do to specify
>> >>> >> that
>> >>> >> person A dialing for example number xxx-xxx-xxxx is the moderator
>> >>> >> (via
>> >>> >> HTTP)? Would I have to create my own call to the system and add say
>> >>> >> an
>> >>> >> entry to DialPlan with the right parameter for the moderator, then
>> >>> >> create the conference?
>> >>> >>
>> >>> >> b) When a conference is created, or when I go to create a new
>> >>> >> conference via HTTP using the API, does it allow for example for
>> >>> >> all
>> >>> >> numbers that will be added to be dialed at once? Or should the
>> >>> >> process
>> >>> >> be dial each participant, sending say 3 http requests via the API?
>> >>> >> The
>> >>> >> API command "conference dial" seems to only take one argument for
>> >>> >> destination number; but I am asking just in case I missed
>> >>> >> something.
>> >>> >>
>> >>> >> Thanks in advance for the help and I apologize for the long email.
>> >>> >>
>> >>> >> Alfredo
>> >>> >>
>> >>> >> _______________________________________________
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>> >>> >> FreeSWITCH-users at lists.freeswitch.org
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>> >>> >>
>> >>> >>
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>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> > MSN:anthony_minessale at hotmail.com
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> > sip:888 at conference.freeswitch.org
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> > googletalk:conf+888 at conference.freeswitch.org
>> >>> > pstn:+19193869900
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
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>> >>> >
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>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
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>> >>>
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>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >>
>> >
>>
>> _______________________________________________
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
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>
>