So as I understand the only thing that I can do is to set up indications.conf. Ok I will try it tomorrow and will write again with my results.
Thanks a lot. 2008/9/4 Eric ManxPower Wieling <[EMAIL PROTECTED]>: > This has nothing to do with the progressinband setting and you should > never use the "r" option. > > eng. Anatoli Marinov wrote: >> Is there any special option which I should enable to activate these tones? >> My progressinband is "yes" and I cal Dial app with "r" option it it right? >> >> >> >> 2008/9/4 Eric ManxPower Wieling <[EMAIL PROTECTED]>: >>> It will do so by default if you have a valid >>> /etc/asterisk/indications.conf (only used for inband tones like after an >>> Answer()) >>> >>> eng. Anatoli Marinov wrote: >>>> Hi guys, >>>> I am trying to configure an asterisk server for our office. >>>> Asterisk 1.4.17 SIP only >>>> >>>> The problem appears when the call comes from external point to our >>>> internal network. So when the server receives the call the channel is >>>> answered and the remote user hears prompt which invite him to enter >>>> internal private number. After that the server starts to wait the >>>> extension. After timeout the server executes Dial application and >>>> sends invite to sip client from our internal network. The problem is >>>> in this point. I want to play ringback tone to remote user when he >>>> waits internal user to pick up his phone but I could not instruct >>>> Asterisk to generate fake ringback in rtp stream . >>> >>> -- >>> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, >>> T-1, PRI, Frame Relay, Linux, and network design. Based near >>> Birmingham, AL. Now accepting clients worldwide. >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> > > -- > Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, > T-1, PRI, Frame Relay, Linux, and network design. Based near > Birmingham, AL. Now accepting clients worldwide. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards eng. Anatoli Marinov _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users