SIPLOGIC.CONF(5)

NAME
SipLogic.conf - SipLogic configuration file for the SvxLink server

DESCRIPTION
svxlink is a general purpose voice service system for ham radio use. This man-page describe the SvxLink server configuration file format for the SipLogic logic core.

CONFIGURATION VARIABLES
Here is the description of all configuration variables that the SipLogic SvxLink logic core understands. The configuration variables are described section for section.

  SipLogic
The SipLogic is used to connect to a Sip-based communication server like Astrisk.

TYPE
    The type for a sip logic core is always Sip.

SIPSERVER
    The ip address or the hostname of the Sip communication server.

SIPPORT
    The udp port that is used for communication (normally 5060).

CALLERNAME
    The name of the station that is displayed instead of the number.

USERNAME
    The username that is used to connect to the Sip communication server.

PASSWORD
    The password for the USERNAME to connect to the Sip communication server.

SIPREGISTRAR
    The sip registrar or better kown as proxy (e.g. "fritz.box" for AVM Fritzboxes)

SIPSCHEMA
    The authentication scheme (e.g. "digest")

SIPEXTENSION
    The extension to used when connecting to a Sip communication server.

AUTOANSWER
    If set the call will be pickuped immediately when received. AUTOANSWER=(0|1)

AUTOCONNECT
    Define a party that will being connected after upon startup of SvxLink. AUTOCONNECT=user123@sipaccount.com

REJECT_INCOMING
    Reject all incoming phone calls where the remote number match this regular expression. For more information on the syntax, seek for REGULAR EXPRESSIONS

ACCEPT_INCOMING
    Only accept incoming phone calls where the remote number match this regular expression. For more information on the syntax, seek for REGULAR EXPRESSIONS

REJECT_OUTGOING
    Reject all outgoing phone calls where the remote number match this regular expression. For more information on the syntax, seek for REGULAR EXPRESSIONS

ACCEPT_OUTGOING
    Only accept outgoing phone calls where the remote number match this regular expression. For more information on the syntax, seek for REGULAR EXPRESSIONS

SEMI_DUPLEX (ToDo)
    If enabled no VOX squelch is used for Sip clients. That means that the audio stream from the SipLogic is sent continuously to the connected SvxLink nodes. That is only meaningful if all connected nodes are repeaters. SEMI_DUPLEX=(0|1)

VOX_FILTER_DEPTH
    see "Local Receiver Section"

VOX_THRESH
    see "Local Receiver Section"

SQL_START_DELAY
    see "Local Receiver Section"

SQL_DELAY
    see "Local Receiver Section"

SQL_HANGTIME
    see "Local Receiver Section"

SIP_LOGLEVEL
    The debug output of the pjsip library (0..6)

JITTER_BUFFER_DELAY
    A jitter buffer is used to prevent gaps in the audio when the network connection do not provide a steady flow of data. Set this configuration variable to the number of milliseconds to buffer before starting to process the audio. Default: 0.

REG_TIMEOUT
    Time in seconds to renewal the registration at the sip server.

CALL_TIMEOUT
    Time in seconds after an outgoing call will be terminate when the party will not pickup the call.

SIP_CTRL_PTY
    Using this configuration variable it is possible to specify a path to a UNIX 98 PTY that allows a dtmf control of each single SvxLink logic. SvxLink will create a softlink to the actual slave PTY. For that reason, SvxLink must have write permissions in the directory where the softlink should be created. Sending commands to the PTY is as simple as doing a echo 'C1#' > /path/to/pty after starting SvxLink. The device works bidirectional, received dtmf characters (from local application) are output via this interface. Example: SIP_CTRL_PTY=/dev/shm/dtmf_ctrl

SIP_PREAMP
    The incoming signal from the Sip network will be amplified by the specified number of dB. This can be used as a last measure if the input audio level can't be set high enough on the analogue side. A value of 6dB will double the signal level. Note that this is a digital amplification. Hence it will reduce the dynamic range of the signal so usage should be avoided if possible. It's always better to correct the audio level before sampling it.

SIP_LIMITER_THRESH
    Set the threshold, in dBFS, for the audio limiter. The audio limiter really is a compressor with a very steep compression ratio like 10:1. The limiter is used to help keeping down the audio level from Sip network for overmodulating stations. Try values from -6 to -12 if the incoming Sip audio is to high. Set to 0 to deactivate it.

PHONENUMBER_TO_TG
    Comma-separated list of assignments of phonenumber(full||substring):Talkgroup. When a phone call is received, the call is assigned to the defined TG according to the recognized number. This can be used to prevent the transfer of an SIP call to a network or to carry out a country-specific assignment. The number format transmitted must match the configuration (+49 != 0049). Whole SIP numbers as well as parts of phone numbers can be configured, with the evaluation taking place on the left-hand side. Example: PHONENUMBER_TO_TG=0049:262,0034:214,0043:232,0049123454543:991

USE_TCP
    If set to 1, TCP is used as transport layer instead of UDP (default).

IGNORE_REGISTRATION
    If set to 1, the SipLogic doesn't check if the incoming caller is registered at the own Sip server (default: false).

FILES

/etc/svxlink/svxlink.d/SipLogic.conf
    The SipLogic specific configuration file.

/etc/svxlink/svxlink.conf
    The system wide configuration file.

~/.svxlink/svxlink.conf
    Per user configuration file.

/etc/svxlink/svxlink.d/*
    Additional configuration files. Typically one configuration file per module.

AUTHOR
Adi Bier (DL1HRC)

SEE ALSO
svxlink.conf(5)