Awesome! That's even nicer. Appreciate it. Alfredo Q-V On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale <anthony.minessale@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@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@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@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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