[Freeswitch-users] Dial multiple users for "conference" call

Wed Jan 6 17:40:45 MSK 2016

My idea was, whenever dialing from my SIP client: 1001;1002;1003, all of
them would be called at the same time - and all of them would join a mutual
call.

On Wed, Jan 6, 2016 at 3:38 PM, Michael Nielsen <mic.niel84 at gmail.com>
wrote:

> I see, so I somehow need to dynamically create my dialplan upon dialing.
>
> On Wed, Jan 6, 2016 at 3:13 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Or better yet, look at the madboss example in our vanilla config!
>>
>>     <!--This extension will start a conference and invite several people
>> upon entering -->
>>
>>     <extension name="mad_boss">
>>
>>       <condition field="destination_number" expression="^0913$">
>>
>>         <!--These params effect the outcalls made once you join-->
>>
>>         <action application="set" data="conference_auto_outcall_caller_id_name=Mad
>> Boss"/>
>>
>>         <action application="set" data=
>> "conference_auto_outcall_caller_id_number=0911"/>
>>
>>         <action application="set" data=
>> "conference_auto_outcall_timeout=60"/>
>>
>>         <action application="set" data=
>> "conference_auto_outcall_flags=none"/>
>>
>>         <!--<action application="set"
>> data="conference_auto_outcall_announce=say:You have been called into an
>> emergency conference"/>-->
>>
>>         <!--Add as many of these as you need, These are the people you
>> are going to call-->
>>
>>         <action application="conference_set_auto_outcall" data=
>> "loopback/9664"/>
>>
>>         <action application="conference" data="madboss3 at default"/>
>>
>>       </condition>
>>
>>     </extension>
>>
>> On Tue, Jan 5, 2016 at 8:01 PM, Bote Man <bote_radio at botecomm.com> wrote:
>>
>>> There exists an API command for the conference called "relate" which can
>>> mute one conferee relative to another specified conferee. If you wish to
>>> keep conferees separate this might help you, but it requires some scripting
>>> to implement at call setup time. Perhaps I don't understand your question
>>> fully?
>>>
>>> Bote
>>>
>>>
>>> On Tue, Jan 5, 2016 at 7:35 AM, Michael Nielsen <mic.niel84 at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Many thanks for the link - I'l have a look at it. But it seems quite
>>>> "difficult" to do. After browsing on the mod_conference, I found this
>>>> example:
>>>>       <action application="conference"
>>>> data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/>
>>>>
>>>> Doesn't that do something similar?
>>>>
>>>> On Tue, Jan 5, 2016 at 1:01 PM, Stanislav Sinyagin <ssinyagin at gmail.com
>>>> > wrote:
>>>>
>>>>> of course you can, you just use mod_conference and some magic.
>>>>>
>>>>> Here's my script which does approximately what you described, but its
>>>>> main purpose is test automation.
>>>>>
>>>>> https://github.com/voxserv/freeswitch-helper-scripts/blob/master/esl/conference_dialer
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 5, 2016 at 11:43 AM, Michael Nielsen <mic.niel84 at gmail.com>
>>>>> wrote:
>>>>> > Is it possible to dial multiple users at once, for initiating a
>>>>> conference
>>>>> > call - without the need for a conference room.
>>>>> >
>>>>> > And without having the client as "connection-point" for all the
>>>>> parties
>>>>> > involved, but having FreeSWITCH handling the merging of calls, so
>>>>> the RTP
>>>>> > stream for each client is similar to a 1-1 call for all parties.
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
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>>>>>
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>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>>
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>>>
>>>
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>>>
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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
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>>
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>>
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>>
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>
>
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