[Freeswitch-users] Requiring Pin of "1" for all conferences + announcement

Fri Jan 24 00:39:08 UTC 2020

Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...

> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Have you tried “+1”?
> 
> Like
> 
> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
> 
> 
> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
> 
> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
> Hello Freeswitch Pros,
> 
> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
> 
> Thank you,
> 
> -Spence
> 
>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>> 
>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>> 
>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>> 
>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
> 
>> 
>> 
>> Thank you for your help on both these related issues.
>> 
>> -Spence
>> 
> 
> _________________________________________________________________________
> 
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> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> -- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
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