eng. Anatoli Marinov
2008-Sep-04 15:16 UTC
[asterisk-users] ringback when the channel is answered
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 . Is there a solution for this? Thanks in advance. -- Best Regards eng. Anatoli Marinov
Eric "ManxPower" Wieling
2008-Sep-04 15:32 UTC
[asterisk-users] ringback when the channel is answered
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.
eng. Anatoli Marinov
2008-Sep-04 17:37 UTC
[asterisk-users] ringback when the channel is answered
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 >-- Best Regards eng. Anatoli Marinov
Eric "ManxPower" Wieling
2008-Sep-04 17:56 UTC
[asterisk-users] ringback when the channel is answered
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.
eng. Anatoli Marinov
2008-Sep-04 19:26 UTC
[asterisk-users] ringback when the channel is answered
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 <eric at fnords.org>:> 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 >-- Best Regards eng. Anatoli Marinov
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 >
eng. Anatoli Marinov
2008-Sep-04 21:00 UTC
[asterisk-users] ringback when the channel is answered
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
eng. Anatoli Marinov
2008-Sep-05 06:35 UTC
[asterisk-users] ringback when the channel is answered
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. >>>>> >>>>> _______________________________________________ >>>>> -- 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