[asterisk-users] ringback when the channel is answered

eng. Anatoli Marinov tolisoft at gmail.com
Fri Sep 5 01:35:57 CDT 2008


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 <tolisoft at gmail.com>:
> 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.
>>>>>
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>>>>
>>>>
>>>>
>>>
>>> --
>>> 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



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