This is a general question on the underlying basic functionality within Asterisk I have been unable to answer.
- An installation of Asterisk with no additional plug-in cards performing any analogue or digital functions, i.e just asterisk on a PC (say a portable).
- LAN based SIP phones
- LAN ATA supporting one analog PSTN line and one analog POT
- All device CODECs set to G711u or G711a
- All devices and asterisk set to DTMF in-band
Does (can) Asterisk in the above scenario provide DTMF decode of in-band DTMF digits prior to AND after a call is established?
To do this would surely require asterisk to be running SOFTWARE G711a/u codecs on all ‘speech packets’ and further a SOFTWARE DTMF decoder on all speech packets for the duration of all calls.
If indeed this is possible and implemented, it would seem to place a high demand on the PC and limit the number of concurrent calls.