Hi Adam,
just a wild guess - try to insert t_reply("180", "Ringing") before
t_relay in your script.On 06/06/2012 10:34 AM, Adam Raszynski wrote: > 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 > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
