[Freeswitch-users] Conference Questions

Fri Jan 15 09:53:48 PST 2010

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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>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
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>> >
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>> >
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>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
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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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