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authorSuper User2007-05-06 15:54:52 +0200
committerSuper User2007-05-06 15:54:52 +0200
commit2ed0fee489c37a6e2d4473f6185ebbe3e746ac11 (patch)
treefcf232bc282c083404cfde0ce5b04236fe202c3e /default
parentfirst commit (diff)
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only for backup, still in coding state - no compile!!!
Diffstat (limited to 'default')
-rw-r--r--default/directory.list21
-rw-r--r--default/h323_gateway.conf13
-rw-r--r--default/interface.conf88
-rw-r--r--default/options.conf165
-rw-r--r--default/routing.conf65
5 files changed, 352 insertions, 0 deletions
diff --git a/default/directory.list b/default/directory.list
new file mode 100644
index 0000000..10f2897
--- /dev/null
+++ b/default/directory.list
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+# The directory translates caller IDs to names.
+# Format: <phone number> <Name>\n");
+
+# The number must be given as received:
+# National number: The national prefix must be included or 'n' must be used.
+# E.g.: n9036681733 or 19036681733 (in case '1' is the national prefix)
+# E.g.: n21250993 or 021250993 (in case '0' is the national prefix)
+# International number: The international prefix must be included or 'i'.
+# E.g.: i4921256483 or 0114921256493 (in case '011' is the intl. prefix)
+# E.g.: i19036681733 or 0019036681733 (in case '00' is the intl. prefix)
+# Subscriber number: No prefix mus be included or 's' must be used.
+# E.g.: s50993 or 1733
+
+# Note: National numbers always require the area code. International numbers
+# always require the country code + area code.
+
+021250993 German number
+0019036681739 American number
+n48416600 German hospital
+
+
diff --git a/default/h323_gateway.conf b/default/h323_gateway.conf
new file mode 100644
index 0000000..96a8f8e
--- /dev/null
+++ b/default/h323_gateway.conf
@@ -0,0 +1,13 @@
+# 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/interface.conf b/default/interface.conf
new file mode 100644
index 0000000..6df206c
--- /dev/null
+++ b/default/interface.conf
@@ -0,0 +1,88 @@
+# interface.conf
+################
+
+
+# Example of an ISDN interface on port 1
+
+#[Ext]
+#port 1
+
+# Example of a multilink ISDN interface (Anlagenanschluss) on port 3
+# and 4
+
+#[Ext]
+#port 3
+#port 4
+
+# Example of an ISDN interface on port 2, which accepts all extensions
+
+#[Int]
+#extension
+#port 2
+
+
+# Example of an ISDN interface on port 2, which accepts only extensions
+# 201, 202 and 203
+
+#[Int]
+#extension
+#msn 201,202,203
+#port 2
+
+
+# Example of three ISDN interfaces on port 2, 3 and 4, which accept
+# extension 201 on all interfaces, and extension 202 and 203 on the first port
+# only.
+# Hint: To make extension ring on all three interfaces parallel, use
+# "interface Int1,Int2,In3" in extension's settings file.
+
+#[Int1]
+#extension
+#msn 201,202,203
+#port 2
+
+#[Int2]
+#extension
+#msn 201
+#port 3
+
+#[Int3]
+#extension
+#msn 201
+#port 4
+
+
+# Example of an ISDN interface on port 1 that requires screening of caller IDs.
+# This is required if the connected line doesn't screen caller IDs.
+# Also this interface will connect bchannel during call setup, so tones are
+# required.
+#[Ext]
+#screen_out unknown 300 national 21250993300
+#screen_out unknown 2* national 212509932*
+#tones yes
+#port 1
+
+
+# Example of an ISDN PRI interface on port 1 that has directed channels.
+# 10 channels (channel 1-10) are incomming only.
+# 10 channels (channel 11-15,17-21) are outgoing only.
+# 10 channels (channel 22-31) are both way.
+# We prefer to use directed channels first, then we request any channel:
+#[Ext]
+#port 1
+#channel force,11,12,13,14,15,17,18,19,20,21,any
+
+
+# Add your interfaces here:
+
+
+[Ext]
+port 1
+
+
+[Int]
+extension
+msn 201,202,203
+port 2
+
+
diff --git a/default/options.conf b/default/options.conf
new file mode 100644
index 0000000..e639350
--- /dev/null
+++ b/default/options.conf
@@ -0,0 +1,165 @@
+# PBX options
+#############
+
+# Turn debugging all on=0xffff or off=0x0000 (default= 0x0000)
+#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
+#define DEBUG_CRYPT 0x1000
+#define DEBUG_ROUTE 0x2000
+#define DEBUG_IDLETIME 0x4000
+#define DEBUG_LOG 0x7fff
+
+#debug 0x0000
+
+# The log file can be used to track actions by the PBX. 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
+
+# Use "alaw" (default) or "ulaw" samples.
+#alaw
+
+# The pbx 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
+# all CPU time forever - your machine hangs.
+#schedule 0
+
+# Use tone sets (default= tones_american).
+# This can be overridden by the extension setting
+#tones_dir tones_american
+
+# Fetch tone sets as specified here.
+# The tone sets will be loaded during startup, and no harddisk access is
+# required. Specify all tone sets seperated by komma.
+# By default, no tone is fetched. Tone sets, that are not specified here, will
+# be streamed from hard disk.
+# Don't use spaces to seperate!
+#fetch_tones tones_american,tones_german,vbox_english,vbox_german
+
+# Extensions directory where all configuration files and messages for all
+# extensions are stored (default= extensions).
+#extensions_dir extensions
+
+# Prefix to dial national call (default= 0).
+# If you omit the prefix, all subscriber numbers are national numbers.
+# (example: Danmark)
+#national 0
+
+# Prefix to dial international call (default= 00).
+# If you omit the prefix, all subscriber numbers are international numbers.
+#international 00
+
+# On external calls, dialing can be done via normal called party number
+# information element or via keypad facility. Some telephone systems require
+# dialing via keypad to enable/disable special functions.
+# 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.
+# If you don't know what to use it for, you don't need it.
+# 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
+# bus/cpu load at all, except when the tone changes.
+# This works only for ISDN ports. It can be overridden by extension's tone set.
+# Defautlt is streaming of tones. Use parameter "american", "german", or
+# "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
+# 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
+
diff --git a/default/routing.conf b/default/routing.conf
new file mode 100644
index 0000000..ff7207f
--- /dev/null
+++ b/default/routing.conf
@@ -0,0 +1,65 @@
+# PBX4Linux routing configuration "routing.conf"
+
+
+# Ruleset: MAIN
+# 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
+
+# Ruleset: EXTERN
+# All calls from external lines are processed here.
+
+[extern]
+dialing=0,1234 : intern extension=200
+dialing=200-299 : intern
+dialing=81 : partyline room=42
+timeout=6 : intern extension=200
+default : disconnect cause=1
+
+
+# Ruleset: INTERN
+# All calls from internal ports are processed here.
+
+[intern]
+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
+dialing=7 enblock : abbrev
+dialing=7 : abbrev select
+dialing=80 : vbox-play
+dialing=81 : partyline room=42
+dialing=90 : powerdial
+dialing=91 : callerid
+dialing=92 : calleridnext
+#dialing=93 : login
+#dialing=94 : powerdial
+#dialing=950 real : callback
+#dialing=953 : forward diversion=cfu
+#dialing=954 : forward diversion=cfb
+#dialing=955 : forward diversion=cfnr delay=26
+#dialing=956 : forward diversion=cfp
+#dialing=957 : forward diversion=cfu dest=vbox
+#dialing=958 : forward diversion=cfb dest=vbox
+#dialing=959 : forward diversion=cfnr dest=vbox delay=20
+#dialing=96 : dtmf
+#dialing=970 : calculator connect
+dialing=99 : test
+default : disconnect cause=1 display="Invalid Code"
+
+
+# Ruleset: VOIP
+# All calls will be forwarded to extension 200
+
+[voip]
+ : intern extension=200
+
+