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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 > > > > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
