Thank you, it worked !!!, all the manuals and instructions I lokked at showed the extension on this particular switch.
Now my problem is another, it gets played before the other party answers because it believes the SIPGW (which is connected to the PSTN using ISDN) answers the call as soon as it is accepted by it although not yet answered. That is a SIP issue, but I don't understand why it happens. Other calls that are normally dialed show ringing. This one with anouncement does not. Regards, Jorge A. -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Eric Wieling aka ManxPower Enviado el: Jueves, 30 de Junio de 2005 05:22 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Dial Option A(file.gsm) Jorge Alayon wrote: > Hello, > > I am trying to let someone know that is being called from a specified > location. > For that, the command: > > exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) > > should let the called person hear Anounce.gsm as soon as he/she answers. > > (Only calls with prefix 107 are given this notice). > > The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this > file in every sound directory, but no luck. > > Has anyone used this feature ? Are there any additional parameters or > restrictions ? You never provide a file extension for those sort of stuff. Asterisk will figure that out. Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce)) -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 _______________________________________________ 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 _______________________________________________ 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
