- Register works in both ways
- STUN works as client
- Authentication to remote endpoints only
- Early audio (183) works in both directions
- Caller ID works in both directions
Note: The implementation is only a small subset of many SIP features.
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 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.
For external calls, the list of interfaces is used to select the first
available/not busy interface. If the interface list is stated with +,
the call is forked to all interfaces.
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.
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.
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.
This is needed for peers that require DTMF messages, rather than tones:
- GSM mobile stations
- Asterisk channel API
modified: apppbx.cpp
modified: apppbx.h
If no specific interface is given for the 'extern' rule in the
routing.conf, or if call is forwarded (settings), then only
interfaces marked with 'extern' flag are used.
You need to set this flag when upgrading to this version. See
default/interface.conf for hint.
modified: README
modified: apppbx.cpp
modified: default/interface.conf
modified: interface.c
modified: interface.h
Fix of conference release bug.
Calls can now be forwarded during alerting phase via "*3#".
modified: README
modified: apppbx.cpp
modified: configure
modified: configure.ac
new file: default/openbsc.cfg
modified: dss1.cpp
modified: gsm.cpp
modified: joinpbx.cpp
-> if initial caller uses pure data mode (or video), the bchannels for this call are handled in HDLC mode. (hardware/software briding is still applicable.)
modified: apppbx.cpp
modified: chan_lcr.c
modified: dss1.cpp
modified: dss1.h
modified: lcradmin.c
modified: lcrsocket.h
modified: mISDN.cpp
modified: mISDN.h
modified: message.h
modified: socket_server.c
Andreas Eversberg
Andreas Eversberg