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authorSuper User2007-07-18 11:44:25 +0200
committerSuper User2007-07-18 11:44:25 +0200
commitbf3c4d173ad6ecf845de2d9e6d11ab87769d0943 (patch)
tree92ebe71b8a6f61c58ba31774bae4e739a1de42a3 /default
parentalpha phase is open, this means: (diff)
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fixes & improvements
modified: admin_client.c modified: apppbx.cpp modified: asterisk_client.c deleted: default/h323_gateway.conf modified: default/options.conf modified: default/routing.conf modified: dss1.cpp modified: message.c modified: todo.txt modified: vbox.cpp modified: vbox.h
Diffstat (limited to 'default')
-rw-r--r--default/h323_gateway.conf13
-rw-r--r--default/options.conf96
-rw-r--r--default/routing.conf10
3 files changed, 12 insertions, 107 deletions
diff --git a/default/h323_gateway.conf b/default/h323_gateway.conf
deleted file mode 100644
index 96a8f8e..0000000
--- a/default/h323_gateway.conf
+++ /dev/null
@@ -1,13 +0,0 @@
-# Gateway configuration for incoming h323 calls
-#
-# Incoming calls will be mapped to the given extensions. If calling host
-# is not listed here, it will be processed as extenal call.
-# IP numbers must be given here, not host names.
-# The "connect" option allows to send back a connect before the call is
-# actually answered. Some H323 client will only allow entering of digits
-# after a connect.
-# Using "dtmf" connects and also enables DTMF detection.
-#
-# <host ip> <extension> [connect | dtmf]
-192.168.0.2 200
-
diff --git a/default/options.conf b/default/options.conf
index e639350..21c8697 100644
--- a/default/options.conf
+++ b/default/options.conf
@@ -1,14 +1,16 @@
-# PBX options
+# LCR options
#############
# Turn debugging all on=0xffff or off=0x0000 (default= 0x0000)
+# Note that debugging is for developer only. If you wan't to 'see the LCR
+# working', you will find a logging feature below. Also detailed traces
+# are possible using the admin tool.
#define DEBUG_CONFIG 0x0001
#define DEBUG_MSG 0x0002
#define DEBUG_STACK 0x0004
#define DEBUG_BCHANNEL 0x0008
#define DEBUG_PORT 0x0100
#define DEBUG_ISDN 0x0110
-#define DEBUG_H323 0x0120
#define DEBUG_VBOX 0x0180
#define DEBUG_EPOINT 0x0200
#define DEBUG_CALL 0x0400
@@ -19,19 +21,19 @@
#debug 0x0000
-# The log file can be used to track actions by the PBX. Omit the parameter
+# The log file can be used to track actions by the LCR. Omit the parameter
# to turn off log file. By default, log file is located inside the directory
-# "/usr/local/pbx/log".
-#log /usr/local/pbx/log
+# "/usr/local/lcr/log".
+#log /usr/local/lcr/log
# Use "alaw" (default) or "ulaw" samples.
#alaw
-# The pbx should run as real time process. Because audio is streamed and
+# The LCR should run as real time process. Because audio is streamed and
# ISDN protocol requires a certain response time, we must have high priority.
# By default, the process runs with realtime scheduling and high priority.
# To debug, it is whise to use "schedule" with no parameter to turn off
-# realtime scheduling. In case of an endless loop bug, PBX4Linux will take
+# realtime scheduling. In case of an endless loop bug, LCR will take
# all CPU time forever - your machine hangs.
#schedule 0
@@ -66,76 +68,6 @@
# By default keypad facility is disabled.
#keypad
-# Internal/external ports (cards connected to your isdn line)
-# MUST be the card number. Use "./pbx query" to list.
-# Add "ptp" for using internal port as point-to-point. (Only required for NT mode ports.)
-# Example: port 2
-# port 3 ptp
-port 2
-port 3
-
-# Specify the H323 endpoint name. If omitted the hostname is used.
-#h323_name PBX4Linux
-
-# Incoming H323 calls can be connected prior answer, because some clients will
-# not play any inband tones during ringing, they just wait as nothing would
-# happen.
-# This only works for external calls. If a H323 caller is authenticated via
-# h323_gateway.conf, a special "connect" option may be used to connect as
-# soon as the call is received.
-# By default this feature is turned off.
-#h323_ringconnect
-
-# Specify which codecs may be used for H323 calls
-# "h323_law" ALaw and muLaw codec which requre more than 64k internet
-# connection cause by overhead. The parameter defines the frame
-# size. The size range is 10 - 240.
-# "h323_g726" The adpcm codec G726. The parameter defines the bits per sample.
-# The bits must be 2, 3, 4, or 5. (16, 24, 32, 40 kbits/s)
-# The given value will always include all modes with lower value.
-# "h323_gsm" GSM0610 and MicrosoftGSM codecs (not compatible with netmeeting)
-# The prameter defines the frame size. The frame range is 1 - 7.
-# "h323_lpc10" Codec with very low bandwith usage which can even be used on
-# slow internetconnections like 9600 kBit (about 300 bytes/s)
-# "h323_speex" Non standard Speex codec. The parameter defines the mode.
-# The mode range is 2 - 6.
-# The given value will always include all modes with lower value.
-# "h323_xspeex" Non standard XiphSpeex codec. The parameter defines the mode.
-# The mode range is 2 - 6.
-# The given value will always include all modes with lower value.
-# The priority of the codecs to use is given by it's order.
-# By default, no codec is used
-h323_gsm 4
-h323_g726 2
-#h323_lpc10
-#h323_speex 4
-#h323_xspeex 4
-h323_law 64
-
-# To allow incoming calls via H323, the following option is used:
-# "h323_icall [<prefix>]"
-# The given prefix is used for incoming calls which do not send any dialing
-# information. If you like to route calls to an extension, give extension
-# dialing as specified at numbering_ext.conf.
-# By default no calls are accepted.
-# Omit the prefix and it must be dialed by the caller.
-h323_icall 0
-
-# Specify the port to listen on incoming H323 connections.
-# The default value is 1720.
-#h323_port 1720
-
-# To register with a gatekeeper, the following option is used:
-# "h323_gatekeeper [<host or ip>]
-# If no parameter is given, the gatekeeper is searched automatically.
-#h323_gatekeeper
-
-# To use dtmf detection for call from or to ISDN, uncomment the keyword "dtmf".
-# By default dtmf detection is used. Note that dtmf detection needs cpu time.
-# Dtmf detection is essential when handling the call after connect using
-# keypad. (conferrence, callback, ect...)
-#nodtmf
-
# For calls to external where caller id is not available, this id is used.
# It is sent of type "subscriber number". This ID is only usefull if the
# external line will not screen caller id. It will be sent anonymous.
@@ -143,12 +75,6 @@ h323_icall 0
# Default is nothing.
#dummyid 0
-# If your external ISDN line(s) support inband patterns prior call connect,
-# you may say 'yes' here. In this case the PBX's tones are used for incoming
-# calls. This may require a special subscription because it can be abused
-# to transfer audio prior charge of call
-#inbandpattern no
-
# Tones/announcements are streamed from user space. It is possible to use
# the module "mISDN_dsp.o" instead. It provides simple tones with much less cpu
# usage. If supported by special hardware, tones are loops that require no
@@ -158,8 +84,8 @@ h323_icall 0
# "oldgerman". "oldgerman" sounds like the old german telephone system (POTS).
#dsptones none
-# Source email address of the PBX. E.g. it is used when sending a mail
+# Source email address of the LCR. E.g. it is used when sending a mail
# from the voice box. It is not the address the mails are sent to.
# Most mail servers require an existing domain in order to accept mails.
-#email pbx@jolly.de
+#email lcr@your.domain
diff --git a/default/routing.conf b/default/routing.conf
index ff7207f..0e2aaac 100644
--- a/default/routing.conf
+++ b/default/routing.conf
@@ -5,7 +5,6 @@
# Calls with different origins will be processed in different rulesets.
[main]
-h323 : goto ruleset=voip
extern : goto ruleset=extern
intern : goto ruleset=intern
: disconnect cause=31
@@ -17,7 +16,7 @@ intern : goto ruleset=intern
dialing=0,1234 : intern extension=200
dialing=200-299 : intern
dialing=81 : partyline room=42
-timeout=6 : intern extension=200
+#timeout=6 : intern extension=200
default : disconnect cause=1
@@ -29,7 +28,6 @@ dialing=0 : extern
dialing=1 : extern capability=digital-unrestricted
dialing=200-299 : intern
dialing=3 : pick
-dialing=4 : h323
dialing=5 : reply
dialing=6 enblock : redial
dialing=6 : redial select
@@ -56,10 +54,4 @@ dialing=99 : test
default : disconnect cause=1 display="Invalid Code"
-# Ruleset: VOIP
-# All calls will be forwarded to extension 200
-
-[voip]
- : intern extension=200
-