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Asterisk sip dtmfmode
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 noteInband DTMF doesn't work reliably unless the codec is ulaw or alaw (G711). Use rfc2833 or info.
The definitions that Asterisk uses for inband is technically inband audio. RFC2833 is technically also an inband method, but often described incorrectly as out-of-band.
- Asterisk DTMF
- Asterisk cmd SipDTMFmode: Change the DTMF type of an outbound SIP call
- SIP DTMF signalling
- Asterisk SIP channels
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