- 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.
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 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.
Since LCR does not put hands on any RTP frame when directly bridged between
OpenBSC and SIP, it will now allow all speech codecs that are commonly supported
by MS and remote SIP endpoint.
It must be noted that OpenBSC must support forwarding the codec types that
MS and remote SIP endpoints support.
Currently LCR negotiates the following codecs for GSM:
- Full Rate
- EFR
- AMR
- Half Rate
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 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.
On a multipoint bus, it is required to assign a channel with no other
alternative allowed. This is required, because an individual phone on the
bus may not choose a different channel, while other phones accept the
indicated channel. Also an individual phone does not have the information
about other available channels. On a point-to-point configuration it is
possible anyway.
In this multipoint case, the channel assignment is now forced automatically,
even if the 'force' keyword is not specified in the "out-channel" of
interface.conf.
Additionally "lcradmin portinfo" shows the channel selection settings now.
Last character of unterminated line was ignored.
Minor bug fix in 2600 Hz pulse dialing.
modified: README
modified: action_vbox.cpp
modified: crypt.cpp
modified: extension.c
modified: gsm_conf.c
modified: interface.c
modified: macro.h
modified: route.c
modified: ss5.cpp
modified: ss5.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
Added limitation option for maximum dialed digits. If dial string exceeds
that limit, overlap-dialing is used to complete dial string.
Siemens EWSD (APS V16) only allows 20 digits at a time.
modified: README
modified: chan_lcr.c
modified: default/interface.conf
modified: dss1.cpp
modified: dss1.h
modified: ie.cpp
modified: interface.c
modified: interface.h
this recovers 'hang' of bchannel if the reply message got lost due to buffer overflows
fixed some minor warnings
modified: Makefile
modified: README
modified: action_efi.cpp
modified: apppbx.cpp
modified: dss1.cpp
modified: interface.c
modified: mISDN.cpp
modified: mISDN.h
modified: message.h
Andreas Eversberg
Andreas Eversberg