[asterisk-users] ringback when the channel is answered

[eng. Anatoli Marinov]

I do not know but I could not set it up. :) bad luck maybe.


2008/9/4 Steve Totaro <stotaro at totarotechnologies.com>:
> 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 <eric at fnords.org> 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 <eric at fnords.org>:
>>>> 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