X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=default%2Foptions.conf;h=6e5ba13fbdb2f9dac26035354b1fa56c3089390e;hp=e63935000a3bc2f93d5e46e799ec36795ec5d889;hb=5463e1b62a39ce417b610584e3d34a8bc30ac15e;hpb=2ed0fee489c37a6e2d4473f6185ebbe3e746ac11 diff --git a/default/options.conf b/default/options.conf index e639350..6e5ba13 100644 --- a/default/options.conf +++ b/default/options.conf @@ -1,42 +1,54 @@ -# 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_GSM 0x0120 +#define DEBUG_SS5 0x0140 #define DEBUG_VBOX 0x0180 +#define DEBUG_SIP 0x10000 #define DEBUG_EPOINT 0x0200 -#define DEBUG_CALL 0x0400 +#define DEBUG_JOIN 0x0400 #define DEBUG_CRYPT 0x1000 #define DEBUG_ROUTE 0x2000 #define DEBUG_IDLETIME 0x4000 -#define DEBUG_LOG 0x7fff +#define DEBUG_LOG 0x7fffff #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. +# By default, the process runs an normal priority, lika most processes do. # 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 # Use tone sets (default= tones_american). -# This can be overridden by the extension setting +# 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. +# Use parameter "american", "german", or "oldgerman". "oldgerman" sounds like +# the old german telephone system used until end of year 1998. +# This can be overridden by the the tones_dir in the interface.conf. +# Both options.conf and interface.conf can be overridden by extension setting. #tones_dir tones_american # Fetch tone sets as specified here. @@ -47,10 +59,6 @@ # 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) @@ -66,76 +74,6 @@ # 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 []" -# 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 [] -# 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. @@ -143,23 +81,40 @@ 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 -# 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 +# 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 + +# Directory to write lock file and admin socket file to. +# If /var/run does not have the rights to run LCR, you may choose /var/tmp +# or any directory with the appropiet rights LCR runs with. +#lock /var/run + +# Change rights of LCR socket, where lcradmin or chan_lcr connects to. +# By default 700 (user only) rights are set. If Asterisk runs with a different +# user, the rights may be changed to all users (777). +# Rights must have 0 in front, if octal values above are used. +#socketrights 0700 + +# Change user of LCR socket, where lcradmin or chan_lcr connects to. +# So: change to asterisk, if you have asterisk run as user "asterisk" e.g. +#socketuser asterisk + +# Change group of LCR socket, where lcradmin or chan_lcr connects to. +# So: change to asterisk, if you have asterisk run in group "asterisk" e.g. +#socketgroup asterisk + +# Enable polling in main loop. +# This feature is temporarily for test purpose. Don't enable it +#polling + +# Two Loopback interfaces for audio transfer between GSM/Asterisk and mISDN. +# The first interface must provide B-channels for each GSM call or channel +# instance, the seond interface links them to LCR. +# Use 30 B-channels unless you need more due to more instances. +# -> Load with: "modprobe mISDN_l1loop pri=1 nchannel=30" +# By default "mISDN_l1loop.1" and "mISDN_l1loop.2" is used. +#loopback-ext mISDN_l1loop.1 +#loopback-lcr mISDN_l1loop.2