summaryrefslogtreecommitdiffstats
path: root/default/options.conf
blob: e63935000a3bc2f93d5e46e799ec36795ec5d889 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
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