* Multiple network instances are now possible to attach multiple networks
* Early audio handling fixed
* Number type can be given from base station (setup / setup confirm)
* Equal callref for different GSM-MS instances are handled correctly
An experimental feature to send and receive an identification over
voice channel.
If a party answers, the ID is transmitted some seconds afterwards.
The calling party listens 30 seconds after receiving an answer message
for the ID.
Add to your extension's settings file:
dov_ident <id string without white spaces>
dov_log /path/to/log/file
dov_type pwm|pcm
dov_level 0|level
'pwm' survives analog transcoding.
'pcm' is fast and will almost not be recognised.
'level' can be used to alter default signal amplitude (100..30000).
- sending clear forward is now forced at any state
- sending signals are queued until last signal has vanished
- timeout while seized/dialing, as well as after busy/clear back
- release of party clears the line (after timeout)
- several minor features and fixes
it is now possible to break the outgoing exchange with:
2600+2400 140ms
0 ms delay
2400 200ms
manually acknowledgement of the answer signal is required then. therefore
the mute feature must be disabled. the delay feature should be used.
The two argument form for AM_INIT_AUTOMAKE is obsolete and
redundant.
Makefile.am:171: warning: 'INCLUDES' is the old name for
'AM_CPPFLAGS' (or '*_CPPFLAGS')
With this patches i see loggings in asterisk cli by enable debugging with
core set debug 1
Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>
This feature was disabled due to locking issues with older Asterisk versions.
RTP bridging only works when interfaces are bridged so we now check
that the interface has been configured as bridge upon seeing rtp-bridge
during parsing of interface.conf.
Note: This change requires that rtp-bridge appears after bridge for
the interface. If this makes your existing configuration fail because
rtp-bridge appears before the bridge parameter then please move the
rtp-bridge line below the bridge line.
The BSC/MS will send a Hello packet that includes the version number,
make LCR verify this version number and close the socket in case it
does not match a supported version.
This is required, if multiple HR calls are made, because HR codec
uses global variables. These global variables are stored after
encoding/decoding and recalled before coding/decoding.
Once a pulse digit is detected, it makes no sense to detect DTMF.
Pulses will create distortion with some phones, causing false
detection of DTMF tones.
Andreas Eversberg