Hi All, Simple scenario: - OpenSIPS as call router to SIP termination provider - I have no control on remote gateways and can't generate early media there
Current situation: - After dialing a number user hears silence until call is routed by my termination provider, call routing to mobile networks sometimes takes 10 or more seconds before RINGING or BUSY response I would like to generate call progress in early media until some meaningful response is generated by termination provider I have local FreeSwitch based media/application server and can use it to generate the tone So the only question is how to route early media to FreeSwitch while making a call and how to disable it when response comes from my provider? Kind Regards
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