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).
The two argument form for AM_INIT_AUTOMAKE is obsolete and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')
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.
otherwise. Note: Autoconf manual says that using $< in ordinary make
rules is not portable, but LCR is Linux specific anyway.
Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>
hardcoded location. This is practically the same as the reverted part of
commit 51655a18 except that $(DESTDIR) *is not* prepended to CC defines;
doing so would break staged installs.
Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>
Call received on an interface can directly be forwarded to a given
destination interface, instead of routing the call through PBX
application. This way calls can be forwarded without going through
route.conf.
Currently only SIP and GSM destinations are supported. Also there
are no tones generated, if one side provides no tones, but the
other wants to receive them.
The keyword "bridge <output interface>" in interface.conf is used.
Without that keyword, incomming calls are handled as usual.
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.
There is no linking of any osmocomBB source code required. In order
to use osmocomBB or OpenBSC, just enable the interface, as described
in defaults/interface.conf. At osmocomBB/mobile or at OpenBSC, just
use the option "-m" to enable the socket interface.
Enable GSM at LCR with "./configure --with-gsm-ms --with-gsm-bs".
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.
No more patch is required, just link openbsc directory to LCR source directory and run "configure":
cd lcr
ln -s path_to_openbsc/openbsc .
modified: Makefile.am
modified: Makefile.in
deleted: bootstrap.c
deleted: bootstrap.h
modified: configure
modified: configure.ac
modified: default/gsm.conf
modified: gsm.cpp
modified: gsm.h
modified: gsm_conf.c