[Freeswitch-users] Conference Setup

[Deepika Yadav]

Thanks Michael, using 9099 directly solved the issue.

On Wed, Apr 20, 2016 at 3:31 AM, Michael Collins <msc at freeswitch.org> wrote:

> The first thing you might do is verify that both of these call legs are
> going into the same conference. Run the script, let the call legs get
> answered, then from fs_cli run:
> conference list
>
> I suspect that you will have each user in his own conference. Check and
> see if you have a conference named "9099". If you do then change the target
> of the originate from "&conference(9099)" to just "9099". The former calls
> the conference app directly while the latter sends the call through the
> dialplan, which is probably what you were trying to do.
>
> -MSC
>
> On Tue, Apr 19, 2016 at 4:55 AM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Hi All,
>>
>> I want to set up a conference initiated from a python esl script.
>> My Dialplans are:
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9099$">
>> <action application="conference"
>> data="radioHealth_${strftime(%Y-%m-%d)}+flags{endconf}"/>
>> </condition>
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9098$">
>> <action application="conference"
>> data="radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
>> </condition>
>>
>> Python script snippets:
>>
>> Calling the first person in unmute mode :
>>
>> freeswitchcon =  ESL.ESLconnection('127.0.0.2', '8021', 'ClueCon')
>>
>> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
>> &conference(9099)"
>>
>> Calling the second persion
>>
>> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
>> &conference(radioHealth_${strftime(%Y-%m-%d)}+flags{mute})"
>>
>> Two outbound calls are created but these are not bridged, I can't hear
>> the voice of first person
>>
>> I tried bridge flags
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9098$">
>> <action application="conference" data="bridge
>> :radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
>> </condition>
>>
>>  not sure if it is the correct way.
>>
>> I also tried calling from SIP internal account :
>>
>> freeswitchcon.api("originate","sofia/internal/1004 at X>X.X.X.X:5080 4446")
>>
>> where 4446 transfers a call to the conference dialplan and called other
>> members from the script to add them in the conference
>>
>> but in this case, I get logs as:
>>
>> switch_channel.c:1055 New Channel loopback/app=voicemail:default X.X.X.X
>> 1004-
>>
>>  loopback/app=voicemail:default X.X.X.X 1004-a setup codec opus/
>>
>> and conference is not setup
>>
>> Any help would be appreciated
>>
>> Regards,
>> Deepiak
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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