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
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