The problem was because my res_indications.so not been loaded. I added it in my modules.conf and now everithing works fine.
Thanks a lot 2008/9/5 eng. Anatoli Marinov <[EMAIL PROTECTED]>: > I do not know but I could not set it up. :) bad luck maybe. > > > 2008/9/4 Steve Totaro <[EMAIL PROTECTED]>: >> Why is it an option if it should "never" be used?..... >> >> Thanks, >> Steve Totaro >> >> On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]> >> wrote: >>> 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 >>> >> >> _______________________________________________ >> -- 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 > -- 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