Thanks Michael, using 9099 directly solved the issue. On Wed, Apr 20, 2016 at 3:31 AM, Michael Collins <msc@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@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@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
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