13 Mar
2016
13 Mar
'16
4:29 p.m.
Sorry if this is a stupid question: I've connected to FreeSwicth using mod_verto, now I would like to route the call to a Genesys via SIP. Is it the right approach? is there any documentation for it? The purpose is to be able to integrate my webRTC solution to many contact center platforms. -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.