look up the originate api instead of conference dial<br><br>originate /dial/string conference:myconf+flags{foo} inline<br><br><br><div class="gmail_quote">On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil <span dir="ltr"><<a href="mailto:lawwton@gmail.com">lawwton@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Anthony:<br>
<br>
I think I spoke too soon. I looked at the example; but I was under the<br>
impression based on your previous comment that I would be able to<br>
invoke the conference dial api command over HTTP and specify the flags<br>
as part of the dial string. The wiki page has the following for the<br>
API call I am thinking:<br>
<br>
dial<br>
<br>
Dial a destination via a specific endpoint (ie. call mom from the conference).<br>
<br>
Usage: conference <confname> dial [{dial string<br>
options}]<endpoint_module_name>/<destination> [<callerid_number><br>
[<callerid_name>]]<br>
<br>
I would like to specify the privilege in this api call. Is that<br>
doable? If not how could I accomplish it? Would I be able to pass the<br>
flags in {dial string options}?<br>
<br>
Thanks in advance for the help,<br>
<br>
Alfredo<br>
<br>
On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil<br>
<div><div></div><div class="h5"><<a href="mailto:lawwton@gmail.com">lawwton@gmail.com</a>> wrote:<br>
> Awesome! That's even nicer.<br>
><br>
> Appreciate it.<br>
><br>
> Alfredo Q-V<br>
><br>
> On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale<br>
> <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>> the flags are set as part of the dial string so you can easily choose that,<br>
>> int the example I told you to look at notice the +flags{} bit at the end of<br>
>> some of the dial strings.<br>
>><br>
>><br>
>> On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil<br>
>> <<a href="mailto:lawwton@gmail.com">lawwton@gmail.com</a>> wrote:<br>
>>><br>
>>> Appreciate the fast response Anthony.<br>
>>><br>
>>> Response or ideas on how to implement a) ?<br>
>>><br>
>>> a) It seemed to me like the way to setup the moderator of the<br>
>>> conference is by setting a parameter in the DialPlan and specifying<br>
>>> based on a condition who the moderator is, say for instance the<br>
>>> destination number. That's fine and it makes sense, however, say that<br>
>>> I am creating a new conference and I want to have 3 participants where<br>
>>> one of them is the moderator. What would I have to do to specify that<br>
>>> person A dialing for example number xxx-xxx-xxxx is the moderator (via<br>
>>> HTTP)? Would I have to create my own call to the system and add say an<br>
>>> entry to DialPlan with the right parameter for the moderator, then<br>
>>> create the conference?<br>
>>><br>
>>> Thanks in advance,<br>
>>><br>
>>> Alfredo Q-V<br>
>>><br>
>>> On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale<br>
>>> <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>>> > look at the "mad boss" extension in the default dialplan<br>
>>> > conf/dialplan/default.xml to see how to craft an all-hands conference.<br>
>>> > otherwise individual calls to originate to send people to the conference<br>
>>> > is<br>
>>> > also ok.<br>
>>> ><br>
>>> ><br>
>>> > On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil<br>
>>> > <<a href="mailto:lawwton@gmail.com">lawwton@gmail.com</a>> wrote:<br>
>>> >><br>
>>> >> Hello:<br>
>>> >><br>
>>> >> I've been using asterisk for a little bit over three years now. A<br>
>>> >> couple of months ago I found out about freeswitch, took a look at it,<br>
>>> >> thought it was interesting and moved on. A few weeks ago, I started<br>
>>> >> looking at a project I've been wanting to work on for quite a while<br>
>>> >> using conferences and started exploring systems and different<br>
>>> >> approaches. Based on the requirements I have, I decided to use<br>
>>> >> freeswitch. It seemed like it had the best support for conferencing so<br>
>>> >> I went for it. According to some documentation I found it also seems<br>
>>> >> to allow for more concurrent calls than asterisk which is an added<br>
>>> >> bonus.<br>
>>> >><br>
>>> >> I got a server ready, installed FC8 on it which is what I have in<br>
>>> >> production now, unpacked freeswitch there and so far it's running<br>
>>> >> beautifully. Very painless process really to get it installed, I was<br>
>>> >> happy to see that. Configuration seems a bit different since it's XML;<br>
>>> >> but being a developer myself I can see many advantages to having done<br>
>>> >> that in the future as the system scales and grows in complexity.<br>
>>> >><br>
>>> >> Sorry for the long introduction, getting to my question now. So ...<br>
>>> >> What I want to be able to do is the following:<br>
>>> >><br>
>>> >> Create and control conferences via the HTTP API. I've been reading a<br>
>>> >> bit for the past two days the documentation and I am becoming more<br>
>>> >> familiar now with how things are done using ESL, the support for PHP,<br>
>>> >> perl and I believe others.<br>
>>> >><br>
>>> >> a) It seemed to me like the way to setup the moderator of the<br>
>>> >> conference is by setting a parameter in the DialPlan and specifying<br>
>>> >> based on a condition who the moderator is, say for instance the<br>
>>> >> destination number. That's fine and it makes sense, however, say that<br>
>>> >> I am creating a new conference and I want to have 3 participants where<br>
>>> >> one of them is the moderator. What would I have to do to specify that<br>
>>> >> person A dialing for example number xxx-xxx-xxxx is the moderator (via<br>
>>> >> HTTP)? Would I have to create my own call to the system and add say an<br>
>>> >> entry to DialPlan with the right parameter for the moderator, then<br>
>>> >> create the conference?<br>
>>> >><br>
>>> >> b) When a conference is created, or when I go to create a new<br>
>>> >> conference via HTTP using the API, does it allow for example for all<br>
>>> >> numbers that will be added to be dialed at once? Or should the process<br>
>>> >> be dial each participant, sending say 3 http requests via the API? The<br>
>>> >> API command "conference dial" seems to only take one argument for<br>
>>> >> destination number; but I am asking just in case I missed something.<br>
>>> >><br>
>>> >> Thanks in advance for the help and I apologize for the long email.<br>
>>> >><br>
>>> >> Alfredo<br>
>>> >><br>
>>> >> _______________________________________________<br>
>>> >> FreeSWITCH-users mailing list<br>
>>> >> <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>
>>> > Anthony Minessale II<br>
>>> ><br>
>>> > FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
>>> > ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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>>> ><br>
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>>> > GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
>>> > IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
>>> ><br>
>>> > FreeSWITCH Developer Conference<br>
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>>> > <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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>>> > pstn:+19193869900<br>
>>> ><br>
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>>> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</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>
>>> <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>
>> Anthony Minessale II<br>
>><br>
>> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
>> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
>> Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>
>><br>
>> AIM: anthm<br>
>> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
>> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
>> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
>><br>
>> FreeSWITCH Developer Conference<br>
>> <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
>> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
>> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
>> pstn:+19193869900<br>
>><br>
>> _______________________________________________<br>
>> FreeSWITCH-users mailing list<br>
>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><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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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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