If the console's station ID is not set, it will be automatically set
when a call is made from the mobile phone or when it performs
inscription to the network. (Only works for cellular networks with
inscription support.)
Also the dtmf encoder will return less samples, if the digit(+pause)
ends, so that the caller call set the next digit to play seamlessly.
A reset function allows to clear the decoder states, to prevent glitches
when re-attaching to an interrupted stream.
Features are:
* Packet based buffer
* Random in, first out
* Adaptive delay compensation (voice)
* Fixed delay (data, optionally MODEM/FAX)
* Interpolation of missing frames
* Any sample size
Command line help shows how many digits and what prefixes can be dialed.
Giving a station ID via command line will be checked for being valid.
The number to call the mobile statione will be checked for being valid.
Prefixes that are defined for a nework will be removed from station ID
automatically.
Multiple station ID lengths are supported:
* C-Netz: 7 or 8 digits, depending on area code length
* A-Netz: 5 or 7 digits; number is truncated to last 5 digits.
* IMTS/MTS: 5 or 7 digits, depending on phone's selector switch.
This way the control channel stays available for other (idle) phones.
No more loss of coverage for other phones, when a call to a mobile is
made.
It is still possible to define a combined control+traffic channel.
(e.g. for single channel setup)
Use -S <dB> for setting RF level or use -S auto for auto level.
When squelch closes, audio is muted. If squelch is closed for some
seconds (depending on network), call is released. (RF loss condition)
The previous loss detection has been removed
This will correct burst errors of received messages. If the message
is too corrupted, it will be ignored, because some element may not
match then. The digits and line signals are checked for consistency,
since they are repeated serveral times in a message.
There is no Supervisory signal during DMS transfer, so there must be
no timeout for Supervisory signal. Instead, the SMS layer will timeout
after some seconds without receiving data.
Paging tones are not created within sound_alsa.c
The audio API is now responsible to switch to paging channel.
In case of SDR, the sdr.c will switch frequency offset.
Sound instance is now called audio instance and uses funcation pointers.
This gives a clean interface to be exchanged with other technologies,
linke SDR.
