[OpenSIPS-Users] How to generate early media until call is established?

[Adam Raszynski]

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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