[Asterisk-Users] custom ring tone

[Fernando Herrera]

If you use FXO you must answer before sending media. If you use BRI, I think
you should be able to send media after sending Progress/Alerting. 

But, watch out!! As trixter said in a previous mail

>> Check your local laws on that

Local regulations may limit the amount of information you can send before
connect. Here in Argentina, you have 1 minute of "wait for answer" time.

Kind regards, 


Fernando Herrera

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de John Hill
Enviado el: Jueves, 22 de Septiembre de 2005 17:25
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] custom ring tone

I was thinking of PSTN over FXO cards. When I see PSTN I think pots.

You mentioned BRI whould PRI do as well?

--john

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Fernando 
> Herrera
> Sent: Thursday, September 22, 2005 3:07 PM
> To: jnovack at stromberg-carlson.org; 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] custom ring tone
> 
>  
> John,
> >> Ringback is provided by your PSTN provider until answer by
> asterisk.
> >> You have no control until you answer
> 
> Generally the ringback tone is sent by the last ClassV/Class IV switch 
> in the telephony path. This is for Telco's to send inband 
> error/progress/information announcements. However, some telcos just 
> send back the relase indicating a certain Release Cause Value and 
> letting you (in case you are another Telco) decide whether to play an 
> announcement or not.
> 
> Marko,
> I think that the DIAL command will match your needs. When you get an 
> incoming call to your asterisk (through any channel, let's say, just 
> as an example, the incoming call comes from an ITSP through a SIP
> channel) you
> configure the Asterisk to send the Music On Hold as a ring back tone 
> (Dial(SIP/1234|90|m)). Though, when you got an incoming call, this 
> will
> happen:
> 
> 1. The ITSP sends an INVITE to your asterisk 2. Asterisk answers with 
> a TRYING 3. Then. Asterisk will send a 183 (Session Progress) and you 
> start transmiting RTP. Normally, you will send the RTP for ring back 
> tone (tuuu tuuu). Here, you will send music on hold through the RTP 
> channel.
> 4. At this very same moment, the asterisk's end user's phone starts 
> ringing.
> 
> 
> 
> You will be able to implement such thing with SIP or H.323 channels if 
> you connect to PSTN through an ITSP. In case your asterisk is 
> connected to PSTN through POTS, you will only be able to do it if you 
> use ISDN.
> If you are
> using FXS/FXO, you won't be able to do it, since in this case the 
> ringback tone is generated by the TELCO's Class V switch.
> 
> Kind regards,
> 
> Fernando Herrera
> 
> -----Mensaje original-----
> De: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] En nombre de John 
> Novack Enviado el: Jueves, 22 de Septiembre de 2005 16:46
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: Re: [Asterisk-Users] custom ring tone
> 
> 
> 
> Marko Rakar wrote:
> 
> >I am not interested in Dial app, I want the callers who are calling 
> >FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, 
> >gsm or whatever)
> >
> >  
> >
> 
> ??
> Ringback is provided by your PSTN provider until answer by asterisk.
> You have no control until you answer
> Then you go to IVR, VM or ??
> 
> John Novack
> 
> >For users within asterisk domain who actually use Dial
> command it does
> >not matter and I know that I can have full control over them
> >
> >
> >
> >  
> >
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