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Asterisk sip dtmfmode
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DTMFmode for SIP client configuration
Choices are inband, rfc2833, info or auto
- inband: This send tones as inband audio within the voice stream. The device that you press the key on generates the DTMF tones. - If the codec is not ulaw or alaw then the DTMF tones will be distorted by the audio compression and will not be recognised. If the phone is set for RFC2833 and asterisk is set for inband then you may not hear anything.
- rfc2833: http://www.ietf.org/rfc/rfc2833.txt This is another inband method, that sends DTMF tones separately as specially encoded RTP packets, distinct from audio packets, but within the same network connection.
- info: See SIP method info and SIP info DTMF or http://www.ietf.org/rfc/rfc2976.txt This is an out-of-band method that sends the DTMF signals within SIP on a separate network connection from the media streams.
- auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to audio DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not available in Asterisk 1.0.x.
Please noteThe definitions of the term 'inband' that Asterisk uses are non-standard. In Asterisk, it means transmission as audio tones, just like speech. RFC2833 is technically also an inband method, but often described incorrectly as out-of-band.
Inband DTMF as audio tone does not work reliably unless the Asterisk codecs setting is ulaw or alaw (G711). The better substitute are 'rfc2833' and 'info'.
- Asterisk DTMF
- Asterisk cmd SipDTMFmode: Change the DTMF type of an outbound SIP call
- SIP DTMF signalling
- Asterisk SIP channels
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