Make tones-dir option available for all interface (interface.conf)
Fixed usage of uninitialized memory, thax to valgrind
Data-Over-Voice 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).
Experimental crypto feature: Support for libvootp
Fixed several compiler warnings
Make LCR compile, even if POTS/FXS is not supported by mISDN Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>
Added option to change DTMF decoding threshold level If not given, the DSP modules' default value is used, rather than setting it to 0. This was a bug.
Fix: Disable DTMF dialing after first received KP (pulse) digit 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.
Add FXS support This requires FXS support to mISDN too.
SIP: Allow early audio on incomming connections at SIP interface In order to provide internal tones, a clock is used to generate chunks of 160 samples. If no tones are provided and if audio is bridged, it is forwarded as usual. In order to provide early audio on SIP trunk, "tones yes" must be set at interface.conf. In order to receive early audio from SIP trunk, "earlyb yes" must be set at interface.conf.
Fix: Allow recording of audio for SIP/remote/GSM interfaces too
Fix: Always keep transmit timer on when mISDN channel is open This way the buffer load is always calculated correctly.
Cleanup: Make interface name be part of Port class
Fix: Process tx-load when briding with jitter buffer disabled
Define prload of mISDN buffer by chan_lcr (required for fax) Use q<ms> option to peload.
Add global variable for Law encoded silence
Removed complete bchannel handling from chan_lcr The remote application interface does not allow any bchannel to be exported or imported. Audio traffic via socket interface is used instead. The joinremote instance became obsolete and is removed. The remote action (routing) became obsolete, use interface.conf instead. The handling of loopback device became obsolete and was removed The chan_lcr does not rely on mISDN anymore, that means: - can be used with GSM and without mISDN at all. - chan_lcr can be used as internal extension of LCR (e.g. SIP phone) (chan_lcr can be handled as any other interface) - no loopback device to be used anymore.
Adding TX-dejitter feature for briged data to mISDN In case there is data bridged to an mISDN port, the TX-dejitter feature is enabled in the kernel, to keep the delay at a minimum.
Fixed audio bridge to mISDN ports Audio must be bridged, even if the call is not connected, but if audio data is already available.
Added bridgin support for GSM and SIP The dependency on mISDN (loopback interface) is completely removed from GSM and SIP interfaces. The built in bridge of LCR now forwards audio data between these interface instances or between these instances and other instances. Additionally both GSM BS and SIP support direct forwarding of RTP traffic between other SIP endpoint and OpenBSC, so no traffic is forwarded by the LCR itself. This is done by forwarding RTP peer informations between these interface instances.