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).
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.
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.
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.
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.
This is required to bridge traffic beween non-mISDN handlers,
such as GSM, SIP and voice box. Also it bridges traffic between
mISDN handlers and non-mISDN handlers. It is the fundamental step
to get rid of mISDN (loop interface) for non-mISDN handlers.
This is required to bridge audio e.g. between SIP and GSM without
using mISDN. There will be no limitations on 'b-channels' anymore.
Still GSM and SIP requires mISDN, but this will be changed later.
With that bridge I cleaned up some code and also removed the
MESSAGE_DATA, which is not required anymore.
This support is just a simple peer-to-peer support for basic calls.
Currently it requires mISDN_l1loop interface, as every non-ISDN
interface does. Later it will be possible to use it without.
The old method was racy, it did use the callback function to deliver the
result, which need special handling because of possible deadlocks.
Now we use a request function which returns the value directely.
The old method is still available, but will get removed soon.
Signed-off-by: Karsten Keil <kkeil@linux-pingi.de>
chan_lcr can be handled as an interface. This way it is possible to (e.g.):
- make a SIP phone become an LCR extension with all LCR features.
- make conference calls. (untested)
- perform parallel ringing. (ISDN phone and SIP phones can ring in
parallel.)
- do voice recoding.
It is still also possible to link chan_lcr directly without interface
(as before).
Documentation/howto for that will follow.
Andreas Eversberg
Andreas Eversberg