Add configuration for vSat

Add sems install and config
Add FS profiles for Vsat incoming calls via sems + related gateway
This commit is contained in:
Keith Whyte 2021-03-11 08:04:26 +01:00
parent 74b07b1fa8
commit ec16a0ddc1
10 changed files with 939 additions and 0 deletions

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@ -0,0 +1,78 @@
# optional path from where to load the profiles if not set mod config path is used
#profiles_path=/etc/sems/etc//profiles/
# profiles - comma-separated list of call profiles to load
#
# <name>.sbcprofile.conf is loaded from module config
# path (the path where this file resides) or from profiles_path if set
profiles=mux-in,mux-out
# active call profile - comma separated list, first non-empty is used
#
# o active_profile=<profile_name> always use <profile_name>
#
# o active_profile=$(ruri.user) use user part of INVITE Request URI
#
# o active_profile=$(paramhdr) use "profile" option in P-App-Param header
#
# o any replacement pattern
#
#active_profile=mux-in
active_profile=$M($Rn_$rU=>ifname_ext),mux-out
# regex_maps - comma-separated list of regex maps to load at startup, for $M()
#
# regex=>value maps for which names are given here are loaded from
# this path, e.g. src_ipmap.conf, ruri_map.conf, usermap.conf
#
#regex_maps=src_ipmap,ruri_map,usermap
regex_maps=ifname_ext
# load_cc_plugins - semicolon-separated list of call-control plugins to load
# here the module names (.so names) must be specified, without .so
# analogous to load_plugins in sems.conf
#
# e.g. load_cc_plugins=cc_pcalls;cc_ctl
#load_cc_plugins=cc_pcalls;cc_ctl
# handle OPTIONS messages in the core? (with limits etc)
# Default: no
core_options_handling=yes
# How many threads to use for processing out-of-dialog messages, default: 1
# out_of_dialog_threads=4
## RFC4028 Session Timer
# default configuration - can be overridden by call profiles
# - enables the session timer ([yes,no]; default: no)
#
#enable_session_timer=yes
# - set the "Session-Expires" parameter for the session timer.
#
# session_expires=240
# - set the "Min-SE" parameter for the session timer.
#
# minimum_timer=90
# session refresh (Session Timer, RFC4028) method
#
# INVITE - use re-INVITE
# UPDATE - use UPDATE
# UPDATE_FALLBACK_INVITE - use UPDATE if indicated in Allow, re-INVITE otherwise
#
# Default: UPDATE_FALLBACK_INVITE
#
#session_refresh_method=UPDATE
# accept_501_reply - accept 501 reply as successful refresh? [yes|no]
#
# Default: yes
#
#accept_501_reply=no

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[Unit]
Description=SEMS Proxy
Requires=sysinit.target system.slice tinc.service
After=network.target
[Service]
Type=simple
WorkingDirectory=/etc/sems/
ExecStart=/usr/sbin/sems -P /var/run/sems/sems.pid -u sems -g sems -f /etc/sems/sems.conf -E
TimeoutStopSec=5
Restart=always
RestartSec=60
[Install]
WantedBy=multi-user.target

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@ -124,6 +124,7 @@ class rhizo_base {
$reg_provider = hiera('rhizo::reg_provider', 'true')
$voip_chans_max = hiera('rhizo::voip_chans_max', '200')
$voip_mod = hiera('rhizo::voip_mod', 'lambda num: num')
$has_vsat = hiera('rhizo::has_vsat', 0)
# Subscription SMS notification
$notice_msg = hiera('rhizo::notice_msg')
@ -197,6 +198,9 @@ class rhizo_base {
if $operatingsystem != 'Debian' {
include rhizo_base::icinga
}
if $has_vsat == 1 {
include rhizo_base::sems
}
if $vpn_ip_address == $riak_ip_address {
if $operatingsystem != 'Debian' {

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@ -0,0 +1,74 @@
# Class: rhizo_base::sems
#
# This module manages the sems package and config
#
# Parameters: none
#
# Actions:
#
# Requires: see Modulefile
#
# Sample Usage:
#
class rhizo_base::sems {
$vpn_ip_address = $rhizo_base::vpn_ip_address
$vsat_ip_address = hiera('rhizo::vsat_ip_address', '127.0.0.1')
$vsat_serv_ip_address = hiera('rhizo::vsat_serv_ip_address', '192.168.12.10')
$rtp_mux_port = hiera('rhizo::rtp_mux_port', '5000')
package { ['sems', ]:
ensure => installed,
require => Class['rhizo_base::apt'],
}
systemd::unit_file { 'sems.service':
source => "puppet:///modules/rhizo_base/systemd/sems.service",
require => Package['sems'],
}
file { '/etc/sems/sems.conf':
content => template('rhizo_base/sems.conf.erb'),
require => Package['sems'],
}
file { '/etc/sems/etc/mux-in.sbcprofile.conf':
content => template('rhizo_base/mux-in.sbcprofile.conf.erb'),
require => Package['sems'],
}
file { '/etc/sems/etc/mux-out.sbcprofile.conf':
content => template('rhizo_base/mux-out.sbcprofile.conf.erb'),
require => Package['sems'],
}
file { '/etc/sems/etc/ifname_ext.conf':
content => template('rhizo_base/ifname_ext.conf.erb'),
require => Package['sems'],
}
file { '/etc/sems/etc/':
ensure => directory,
source => 'puppet:///modules/rhizo_base/sems-etc',
owner => 'root',
recurse => remote,
purge => false,
}
file { '/etc/freeswitch/sip_profiles/vsat.xml':
content => template('rhizo_base/vsat.xml.erb'),
require => Package['freeswitch'],
}
file {'/etc/freeswitch/sip_profiles/vsat':
ensure => directory,
require => File['/etc/freeswitch'],
}
file { '/etc/freeswitch/sip_profiles/vsat/sems.xml':
content => template('rhizo_base/sems.xml.erb'),
require => Package['freeswitch'],
}
}

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@ -0,0 +1,3 @@
^.*_9519800748$=>mux-in
^.*_9519800756$=>mux-in
^rhizomatica_00.*$=>mux-out

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next_hop=<%= @vpn_ip_address %>:5091
RURI=sip:$rU@<%= @vpn_ip_address %>
a_rtp_mux_ip=<%= @vsat_serv_ip_address %>
a_rtp_mux_port=<%= @rtp_mux_port %>
enable_rtprelay=yes
rtprelay_force_symmetric_rtp=no
# RTP interface to use for A leg
aleg_rtprelay_interface=vsat
# RTP interface to use for B leg
rtprelay_interface=rhizo

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@ -0,0 +1,13 @@
next_hop=<%= @vsat_serv_ip_address %>:5090
RURI=sip:$rU@<%= @vsat_serv_ip_address %>
b_rtp_mux_ip==<%= @vsat_serv_ip_address %>
b_rtp_mux_port=<%= @rtp_mux_port %>
enable_rtprelay=yes
# RTP interface to use for A leg
aleg_rtprelay_interface=rhizo
# RTP interface to use for B leg
rtprelay_interface=vsat

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@ -0,0 +1,682 @@
# $Id$
#
# sems.conf.sample
#
# Sip Express Media Server (sems)
#
# sample configuration file
#
#
# whitespaces (spaces and tabs) are ignored
# comments start with a "#" and may be used inline
#
# example: option=value # i like this option
#
# @filename includes mod_config_path/filename
# @/absolute/path/to/file includes file
############################################################
# Network configuration
# If only one signaling (SIP) and media (RTP) interface is
# to be used, configure the options
# sip_ip / sip_port / media_ip/ rtp_low_port /
# rtp_high_port / public_ip / sig_sock_opts
# If more than one interface is to be used, configure the option
# 'interfaces' and those options for each interface.
############# configuration for single interface ############
# optional parameter: sip_ip=<ip_address>|<device>
#
# - SIP IP where the SIP stack is bound to. This also sets
# the value used for contact header in outgoing calls and
# registrations.
# - If neither 'media_ip' nor 'sip_ip' are set, defaults
# to first non-loopback interface, and the port configured below
# is ignored.
#
# Example:
# sip_ip=10.0.0.34
# sip_ip=en0
# optional parameter: sip_port=<port_number>
#
# - port where its SIP stack should be bound to, ignored if sip_ip not set.
# default: 5060
#
#sip_port=5080
# optional parameter: media_ip=<ip_address>|<device>
#
# - IP address or interface that SEMS uses to send and receive media.
# - If neither 'media_ip' nor 'sip_ip' are set, defaults
# to first non-loopback interface. If 'sip_ip' is set,
# 'media_ip' defaults to 'sip_ip.
#
# Examples:
# media_ip=10.0.0.34
# media_ip=eth0
# optional parameter: rtp_low_port=<port>
#
# - sets lowest for RTP used port (Default: 1024)
rtp_low_port=10000
# optional parameter: rtp_high_port=<port>
#
# - sets highest for RTP used port (Default: 0xffff)
rtp_high_port=60000
# optional parameter: public_ip=<ip_address>
#
# - near end NAT traversal. when running SEMS behind certain static
# NATs, use this parameter to inform SEMS of its public IP
# address. If this parameter is set, SEMS will write this value
# into SDP bodies and Contact.
# If this parameter is not set, the local IP address is used.
# N.B., there is no support for port translation; the local
# RTP port is advertised in SDP in either case.
#
# Example:
# public_ip=75.101.219.48
#
# optional parameter: sig_sock_opts=option,option,option,...
#
# - signaling socket options
# o force_via_address - force sending replies to 1st Via
# o no_transport_in_contact - do not add transport to contact in replies
#
# Example:
# sig_sock_opts=force_via_address,no_transport_in_contact
# optional parameter: tcp_connect_timeout=<timeout in millisec>
# Default: 2000 (2 sec)
# optional parameter: tcp_idle_timeout=<timeout in millisec>
# Default: 3600000 (1 hour)
############# configuration for multiple interfaces ############
# interfaces = <list of interface names>
#
# 'interfaces' must be set if more than one interface is to be used
# for the same purpose (e.g. more than one interface for SIP).
# Configure additional interfaces if networks should be bridged or
# separate networks should be served.
#
# For each interface, a set of parameters suffixed with the
# interface name should be configured.
#
# Please note that for each additional interface,
# 'sip_ip_[if_name]' is mandatory (but can be the interface
# name, then the first assigned IP is used). The other
# parameters are optional. 'media_ip_[if_name]'
# is derived from 'sip_ip_[if_name]' if not set.
# 'public_ip_[ip_name]' is also based on 'sip_ip_[if_name]'
# if not set explicitly.
#
# Example:
interfaces=rhizo,vsat
#
sip_ip_rhizo=<%= @vpn_ip_address %>
sip_port_rhizo=5095
media_ip_rhizo=<%= @vpn_ip_address %>
rtp_low_port_rhizo=12000
rtp_high_port_rhizo=13000
#
sip_ip_vsat=<%= @vsat_ip_address %>
sip_port_vsat=5090
media_ip_vsat=<%= @vsat_ip_address %>
rtp_low_port_vsat=14000
rtp_high_port_vsat=15000
#sig_sock_opts_vsat=force_via_address
tcp_connect_timeout_vsat=1000
tcp_idle_timeout_vsat=900000
############# other network configuration ##############################
# NAT handling for SIP:sip_nat_handling={yes|no}
#
# Learn remote next hop address from the source of the address where
# requests are received from. This option does not apply to the sbc module.
#
# default: no
#
#sip_nat_handling=yes
# NAT handling for RTP: force_symmetric_rtp={yes|no}
#
# Force comedia style "symmetric RTP" NAT handling, i.e.
# learn remote RTP address from where RTP packets come from
# (This option does not apply to the sbc module's RTP relay)
#
# default: no
#
force_symmetric_rtp=no
# optional parameter: outbound_proxy=uri
#
# - this sets an outbound proxy for dialogs and registrations initiated
# by SEMS. A preloaded Route header containing the uri is added to
# each initial request. The request is then sent to destination obtained
# by resolving the uri. If outbound_proxy is not set (default setting),
# no preloaded Route header is added and request is sent to destination
# obtained by resolving r-uri. Resolving is done by SIP stack with DNS
# if uri contains domain name. Warning: If uri can not be resolved, no
# requests will be sent out at all!
#
# default: empty
#
# Example:
# outbound_proxy=sip:proxy.mydomain.net
# optional parameter: force_outbound_proxy={yes|no}
#
# - forces SEMS to set outbound_proxy for any requests (not just for
# registrations and dialog initiating requests). See above what setting
# of outbound_proxy means. This option will only have an effect if the
# outbound_proxy option has been set, and it will break 3261 compatibility
# in some cases; better use next_hop.
#
# default: no
#
# Example:
# force_outbound_proxy=yes
# optional parameter: next_hop=address[:port][/transport]
# - if this is set, all outgoing requests will be sent to
# this address (IP address or domain name), regardless of R-URI etc.
#
# Examples:
# next_hop=192.168.5.1
# next_hop=foo.example.com:5080/tcp
# optional parameter: next_hop_1st_req={yes|no}
# - if set to yes, next_hop behavior (routing without pre-loaded route set)
# applies only to initial request. Subsequent requests are routed
# normally based on route set learned from reply to initial request.
#
# default: no
#
# Example:
# next_hop_1st_req=yes
# optional parameter:next_hop_for_replies
# - use next_hop for replies, too?
#
#next_hop_for_replies=yes
# RTP multiplexing configuration - see doc/rtp_mux/
#
# IP:port to open an incoming (listening) RTP MUX port
rtp_mux_ip=<%= @vsat_ip_address %>
rtp_mux_port=<%= @rtp_mux_port %>
#
# Interface to send outgoing RTP MUX packets from (port is auto-assigned)
rtp_mux_out_interface=vsat
#
# Parameters for outgoing mux:
# MTU threshold: frame is sent if over this threshold; leave 1 RTP payload size to actual MTU
rtp_mux_mtu_threshold=135
# max frame age: .. or this old (millisec), i.e. max delay introduced
rtp_mux_max_frame_age_ms=30
#
# For gatewaying, additionally a_rtp_mux_ip/a_rtp_mux_port or b_rtp_mux_ip/
# b_rtp_mux_port must be configured in the SBC profile.
#
############################################################
# modules and application configuration
#
# Configuration of plugin (module) loading:
# - if load_plugins is set, only those are loaded.
# - if load_plugins is not set, all modules in the plugin_path
# directory are loaded, except those which are listed
# in exclude_plugins.
#
# optional parameter: plugin_path=<path>
#
# - sets the path to the plug-ins' binaries
# - may be absolute or relative to CWD
plugin_path=/usr/lib/sems/plug-in/
# optional parameter: load_plugins=<modules list>
#
# semicolon-separated list of modules to load.
# If empty, all modules in plugin_path are loaded.
#
# example for announcement with only g711 and ilbc codecs
# load_plugins=wav;ilbc;announcement
load_plugins=wav;isac;l16;speex;g722;gsm;ilbc;stats;sbc;xmlrpc2di
# optional parameter: exclude_plugins=<modules list>
#
# semicolon-separated list of modules to exclude from loading
# ('blacklist'). If empty, all modules in plugin_path are loaded.
# This has only effect it load_plugins is not set.
#
# o precoded_announce: no precoded sample files present
# o py_sems: conflicts with ivr (in some cases)
# o db_reg_agent: needs DB tables
# o cc_* : sbc call control modules (loaded from sbc.conf)
exclude_plugins=precoded_announce;py_sems;db_reg_agent;cc_call_timer;cc_ctl;cc_pcalls;cc_prepaid;cc_prepaid_xmlrpc;cc_rest;cc_syslog_cdr
# optional: load_plugins_rtld_global=<modules list>
#
# load these plugins with RTLD_GLOBAL (by default py_sems,
# dsm, ivr, sbc, diameter_client, registrar_client, uac_auth)
#
#load_plugins_rtld_global=myapp
# optional parameter: application
#
# This controls which application is to be executed for incoming calls
# is no explicit application requested (outgoing calls)
#
# This can be one of
# $(ruri.user) - user part of ruri is taken as application,
# e.g. sip:announcement@host
# $(ruri.param) - uri parameter "app", e.g.
# sip:joe@host.net;app=announcement
# $(apphdr) - the value of the P-App-Name header is used
#
# $(mapping) - regex=>application mapping is read from
# app_mapping.conf (see app_mapping.conf)
# <application name> - application name configured here, e.g.
# application=announcement
#
# examples:
# application = conference
# application = $(mapping)
# application = $(ruri.user)
# application = $(ruri.param)
# application = $(apphdr)
application=sbc
# parameter: plugin_config_path=<path>
#
# - in this path configuration files of the applications
# (e.g. announcement.conf) are searched
plugin_config_path=/etc/sems/etc/
# optional parameter: exclude_payloads=<payload list>
#
# semicolon-separated list of payloads to exclude from loading
# ('blacklist').
#
# For example, to only use low bandwidth codecs:
# exclude_payloads=PCMU;PCMA;G726-40;G726-32;G721;L16
# or, to use only codecs which are not CPU-intensive:
# exclude_payloads=iLBC;speex;
# only use G711 (exclude everything else):
# exclude_payloads=iLBC;speex;G726-40;G726-32;G721;G726-24;G726-16;GSM;L16
############################################################
# logging and running
# optional parameter: fork={yes|no}
#
# - specifies if sems should run in daemon mode (background)
# (fork=no is the same as -E)
fork=yes
# optional parameter: stderr={yes|no}
#
# - debug mode: do not fork and log to stderr
# (stderr=yes is the same as -E)
stderr=no
# optional parameter: loglevel={0|1|2|3}
#
# - sets log level (error=0, warning=1, info=2, debug=3)
# (same as -D)
loglevel=2
# optional parameter: syslog_facility={DAEMON|USER|LOCAL[0-7]}
#
# - sets the log facility that is used for syslog. Using this,
# the log can for example be filtered into a special file
# by the syslog daemon.
#
# Default: DAEMON
#
# Example:
# syslog_facility=LOCAL0
# optional parameter: log_sessions=[yes|no]
#
# Default: no
#
# If log_sessions=yes is set, INFO level log messages are generated
# for each session when it is started and stopped.
#
# log_sessions=yes
# optional parameter: log_events=[yes|no]
#
# Default: no
#
# If log_eventy=yes is set, generic DBG level log messages are
# generated for each event that is posted into an event queue.
#
# log_events=yes
# optional parameter: max_shutdown_time=<time in seconds>
#
# Limit on server shutdown time (time to send/resend BYE
# to active calls). 0 to disable (infinite).
#
# Default: 10
#
#max_shutdown_time = 10
# optional parameter: shutdown_mode_reply="<code> <reason>"
#
# Error reply that is used as reply to INVITE and OPTION
# when SEMS is shutting down.
#
# Default: shutdown_mode_reply="503 Server shutting down"
# optional parameter: cps_limit=<limit>;<err code>;<err reason>
#
# - this sets a maximum calls per sec limit. If that limit is
# reached, no further calls are accepted, but the error reply
# with err code/err reason is sent out.
#
# Default: 0 (None)
#
# Example:
# cps_limit="100;503;Server overload"
###########################################################
# if build with ZRTP support (see Makefile.defs)
# enable ZRTP support in endpoint calls:
#enable_zrtp=yes (default: yes)
#
# enable ZRTP debug log? (prints lots of info)
#enable_zrtp_debuglog=no (default: yes)
############################################################
# tuning
# optional parameter: session_processor_threads=<num_value>
#
# - controls how many threads should be created that
# process the application logic and in-dialog signaling.
# This is only available if compiled with threadpool support!
# (set USE_THREADPOOL in Makefile.defs)
# Defaults to 10
#
# session_processor_threads=50
# optional parameter: media_processor_threads=<num_value>
#
# - controls how many threads should be created that
# process media - on single-processor systems set this
# parameter to 1 (default), on MP systems to a higher
# value
#
# media_processor_threads=1
# optional parameter: rtp_receiver_threads=<num_value>
#
# - controls how many threads should be created that
# receive and relay RTP media - on single-processor systems set this
# parameter to 1 (default), on MP systems to a higher value.
#
# rtp_receiver_threads=1
# optional parameter: session_limit=<limit>;<err code>;<err reason>
#
# - this sets a maximum active session limit. If that limit is
# reached, no further calls are accepted, but the error reply
# with err code/err reason is sent out.
#
# Default: 0 (None)
#
# Example:
# session_limit="1000;503;Server overload"
# optional parameter: options_session_limit=<limit>;<err code>;<err reason>
#
# - this sets a custom response to OPTIONS, if the session count reaches
# a certain limit. This way health monitor could raise an alarm to syste
# administrator.
#
# Default: 0 (None)
#
# Example:
# options_session_limit="900;503;Warning, server soon overloaded"
# optional parameter: dead_rtp_time=<unsigned int>
#
# - if != 0, after this time (in seconds) of no RTP
# a session is considered dead and stopped. If set
# to 0 no check is done for rtp timeout.
#
# default=300 (5 minutes)
#
# Examples:
# # disable RTP timeout
# dead_rtp_time=0
# # RTP timeout after 10 seconds
# dead_rtp_time=10
# optional parameter: use_default_signature={yes|no}
#
# - use a Server/User-Agent header with the SEMS server
# signature and version.
#
# default=no
#
use_default_signature=yes
# optional parameter: signature=<signature string>
#
# - use a Server/User-Agent header with a custom user agent
# signature. Overridden by default signature if
# use_default_signature is set.
#
# signature="SEMS media server 1.0"
# optional parameter: single_codec_in_ok={yes|no}
#
# - use single codec in 200 OK response
#
# default=no
#
# single_codec_in_ok=no
# optional parameter: codec_order=codec_name_1,codec_name2,...
#
# - Codec order used when sending INVITE requests. Codecs in codec_order
# will be on the top of the list followed by other supported codecs
# (if any).
#
# default=empty
#
# codec_order=iLBC,GSM
# optional parameter: ignore_rtpxheaders={yes|no}
#
# - if this is set to yes, RTP extension headers (e.g. when using ZRTP)
# are ignored. If set to no, the whole RTP packets with extension
# headers will be ignored and a debug message is printed on every
# received packet.
#
# default=no
#
# ignore_rtpxheaders=yes
# optional parameter: dtmf_detector={spandsp|internal}
#
# sets inband DTMF detector to use. spandsp support must be compiled in
# for this to have effect if dtmf_detector=spandsp.
#
# default: internal
#
# dtmf_detector=spandsp
# optional parameter: unhandled_reply_loglevel={error|warn|info|debug|no}
#
# the default application logic implemented in the applications is to stop
# the session right after sending BYE, without waiting for a reply. this
# leads to many log entries of the form
# ERROR: [b6fa6bb0] handleSipMsg (AmSipDispatcher.cpp:48): unhandled
# reply: [code:200;phrase:[OK];... ]
#
# This parameter sets the log lovel of unhandled positive (200 class) replies.
#
# default: error
#
# unhandled_reply_loglevel=info
############################################################
# SIP stack settings
# default settings (i.e. leave out) for these should be OK
# for most applications
# Force use of the selected interface? (Default: no)
#
# Use IP_PKTINFO to force use of the selected interface for sending
# packets. Useful in situations with overlapping IP address spaces, when
# the kernel routing table should not be used to determine the sending
# interface.
#
# force_outbound_if=yes
# SIP timers configuration (in milliseconds)
#
# sip_timer_A=<n millisec>
# sip_timer_B=<n millisec>
# ...
# sip_timer_M=<n millisec>
#
# timers A to J as in RFC3261.
# Timer L: handle 200 ACKs automatically in INVITE client trans.
# Timer M: cycle throught multiple addresses in case the R-URI
# resolves to multiple addresses
#
# Warning: Timer values are not checked whether they are appropriate!
# Leave to default values if in doubt.
#
# Example:
# # equivalent to fr_timer=20 in sip-router
# sip_timer_b=20000
# sip_timer_f=20000
# sip_timer_t2=<n millisec) T2 timer configuration
# (Cap for re-send request/response backoff)
#
# Warning: Timer values are not checked whether they are appropriate!
# Leave to default values if in doubt.
#
#sip_timer_t2=4000
# skip DNS SRV lookup? [yes, no]
#
# according to RFC, if no port is specified, destination IP address
# should be resolved with a DNS SRV lookup. If SEMS should not do that
# (only an A record lookup), set disable_dns_srv=yes.
#
# Default: no
#
#disable_dns_srv=yes
# support 100rel (PRACK) extension (RFC3262)? [disabled|supported|require]
#
# disabled - disable support for 100rel
# supported - support it if remote end does, default
# require - required
#
# Default: supported
#
#100rel=require
# force the use of outbound interface? [yes, no]
#
# Useful in case of overlapping networks, or if OS routing can/should not be used.
# Default: no
#
# force_outbound_if=yes
# use raw sockets for sending? [yes, no]
# faster, requires root or CAP_NET_RAW
#
# Default: no
#
# use_raw_sockets=yes
#
# accept forked dialogs on UAS side? [yes|no]
#
# no - INVITE with existing callid+remote_tag is replied with 482.
# yes - INVITE with existing callid+remote_tag+via_branch is replied with 482.
# Forked INVITEs (!= via-branch) are accepted.
#
# Default: yes
#
#accept_forked_dialogs=no
# Make SIP authenticated requests sticky to the proxy? [yes | no]
#
# If enabled, host of request-URI of out-of-dialog requests that are
# authenticated with SIP auth is changed to the previously resolved
# next-hop IP:port.
#
# default: no
#
# proxy_sticky_auth=yes
# Ignore too low CSeq for NOTIFYs? [yes | no]
#
# May be necessary to interwork with simplistic/old SIP event notification
# implementations.
#
#ignore_notify_lower_cseq=yes
#
# Accept final replies without To-tag? [yes|no]
#
accept_fr_without_totag=yes
#
# Log raw messages? [no|debug|info|warn|error]
#
# Default: debug
#
#log_raw_messages=no
#
# Log parsed received messages? [yes|no]
#
# Default: yes
#
#log_parsed_messages=no
# SIP UDP socket receive buffer size (in bytes)
#
# if not set, system default is used (which usually
# is modest). set sytem wide upper limit with
# e.g. sysctl -w net.core.rmem_max=8388608
#
# udp_rcvbuf = <value>
# Number of SIP UDP receiver threads
#
# Default: 4
#
# sip_server_threads=8
# dump conference streams - experimental
# play with: $play -r <samplerate> -c 1 /tmp/123_1_nnnn.s16
# where <samplerate> is in /tmp/123_1_nnnn.s16.samplerate
# (if it hasn't changed in-between...)
#dump_conference_streams=true
#dump_conference_path=/tmp/

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<include>
<gateway name="sems">
<param name="username" value="rhizomatica"/>
<param name="from-user" value="rhizomatica"/>
<param name="password" value="n-a"/>
<param name="proxy" value="<%= @vpn_ip_address %>:5095"/>
<param name="register" value="false"/>
<param name="caller-id-in-from" value="true"/>
</gateway>
</include>

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<profile name="vsat">
<gateways>
<X-PRE-PROCESS cmd="include" data="vsat/*.xml"/>
</gateways>
<settings>
<!-- param name="auto-jitterbuffer-msec" value="100"/ -->
<param name="minimum-session-expires" value="90"/>
<param name="enable-time" value="false"/>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5094"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="OPUS@8000h@20i,G729"/>
<param name="outbound-codec-prefs" value="OPUS@8000h@20i,G729"/>
<param name="rtp-timer-name" value="soft"/>
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="false"/>
<param name="nat-options-ping" value="false"/>
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="inbound-late-negotiation" value="true"/>
<param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
<param name="ext-sip-ip" value="$${local_ip_v4}"/>
<param name="rtp-timeout-sec" value="90"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>