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 _______________________________________________ -- 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
