On 11/9/06, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: > Erick Perez wrote: > >> > > > > > > then why the slin conversion? > > isn't it required when asterisk hears DTMF tones and act acordingly in > > the voicemail? > > same for IVR? > > > > > > Asterisk only needs to process audio streams for DTMF tones (and > transcode) when the DTMF is inband, which is not possible with g729 (or > any other codec besides ulaw/alaw). If you are using IAX, the DTMF is > never inband. If you are using SIP rfc2833, the DTMF tones are > transported in the RTP stream separate from the audio as separate > events, which Asterisk can process while not being able to transcode the > audio stream. If you are using SIP INFO, it uses SIP signaling and has > nothing to do with the audio at all. :) > > Either way, you don't need to transcode to process DTMF. > > -- > Kristian Kielhofner > _______________________________________________ > Astlinux-users mailing list > [email protected] > http://lists.kriscompanies.com/mailman/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to [EMAIL > PROTECTED] >
I have 35 sip hardphones in g729, my voicemail in g729, my ivr in g729 my moh in g729, only 8 zap channels and 3 sip voip providers in g729. hehehe I have payed digium for g729 licences when it wasn't needed.. ;)) (seems like 8 licences were needed instead of 43) thanks Kristian, -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------ _______________________________________________ Astlinux-users mailing list [email protected] http://lists.kriscompanies.com/mailman/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED]
