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tg2sip en Debian 12, Asterisk 22 y FreePBX 17
Raw
gistfile1.txt
# Debian 12 Bookwrom
# Freepbx / Asterisk installation https://gist.github.com/razametal/18785923be942af78ef8a0984903d5dd
# Enable slin48 (for L16@48000) and opus (for OPUS@48000) in Freepbx -> Settings -> Asterisk Sip Settings
apt-get install -y --no-install-recommends build-essential git wget ca-certificates pkg-config libopus-dev libssl-dev zlib1g-dev gperf ccache make php-cli cmake clang libc++-dev libc++abi-dev libsqlite3-dev libpulse-dev
# libopenssl1.1
cd /usr/src
wget http://security.debian.org/debian-security/pool/updates/main/o/openssl/openssl_1.1.1w.orig.tar.gz
tar zvxf openssl_1.1.1w.orig.tar.gz
cd openssl-1.1.1w/
./config
make
make test
make install
mv /lib/x86_64-linux-gnu/libssl.so /lib/x86_64-linux-gnu/libssl.so.3.tmp
# Opus
cd /usr/src
wget https://ftp.osuosl.org/pub/xiph/releases/opus/opus-1.5.tar.gz
tar zvxf opus-1.5.tar.gz
cd opus1.5
./configure
make
make check
make install
# PJSip v2.9
cd /usr/src
wget https://github.com/pjsip/pjproject/archive/refs/tags/2.9.tar.gz
tar -zxvf 2.9.tar.gz
cd pjproject*
wget https://raw.githubusercontent.com/Infactum/tg2sip/master/buildenv/config_site.h -O pjlib/include/pj/config_site.h
./configure
make dep
make
make install
mv /lib/x86_64-linux-gnu/libssl.so.3.tmp /lib/x86_64-linux-gnu/libssl.so
# spdlog 0.17 https://github.com/gabime/spdlog/releases
cd /usr/src
wget https://github.com/gabime/spdlog/archive/refs/tags/v0.17.0.tar.gz
tar -zxvf v0.17.0.tar.gz
cd spdlog*
mkdir build
cd build
cmake ../
make
make install
# tdlib https://github.com/tdlib/td
cd /usr/src
git clone https://github.com/tdlib/td.git
cd td
git reset --hard 21e5ce0 # version 1.8.37
rm -rf build
mkdir build
cd build
cmake -DCMAKE_BUILD_TYPE=Release -DCMAKE_INSTALL_PREFIX:PATH=/usr/local ..
cmake --build . --target install
# tg2sip
cd /usr/src
#wget https://github.com/Infactum/tg2sip/archive/refs/tags/v1.3.0.tar.gz
#tar -zxf v1.3.0.tar.gz
#cd tg2sip*
git clone https://github.com/Lifemotion/tg2sip
cd tg2sip
mkdir build
cd build
cmake -DCMAKE_BUILD_TYPE=Release ../
cmake --build .
mkdir /etc/tg2sip
mkdir /var/tg2sip
cp build/{tg2sip,gen_db} /usr/local/bin
cat >> /etc/tg2sip/settings.ini << EOF
# Tg2SIP setting
[logging]
core=2
tgvoip=5
pjsip=2
sip_messages=true
console_min_level=0
file_min_level=0
;tdlib=3
[sip]
public_address=0.0.0.0
port=5060
id_uri=sip:tg2sip.midominio.ec:5060
callback_uri=sip:[email protected]
raw_pcm=false
thread_count=1
[telegram]
api_id=12345678
api_hash=aabbccdd0eeff0011223344556677880
system_language_code=es
[other]
;extra_wait_time=30
;peer_flood_time=86400
EOF
cat >> /etc/systemd/system/tg2sip.service << EOF
[Unit]
Description=Tg2SIP Gateway
After=network-online.target
Wants=network-online.target
[Service]
Type=simple
# set TG2SIP_STANDARD_FOLDER to yes
# settings -> /etc/tg2sip/settings.ini
# tg_directory -> /var/tg2sip
Environment=TG2SIP_STANDARD_FOLDER=YES
ExecStart=/usr/local/bin/tg2sip
ExecReload=/bin/kill -s HUP $MAINPID
TimeoutStartSec=0
KillMode=process
Restart=on-failure
StartLimitBurst=3
StartLimitInterval=60s
[Install]
WantedBy=multi-user.target
EOF
systemctl daemon-reload
systemctl enable tg2sip.service
#Iniciar sersion en Telegram
TG2SIP_STANDARD_FOLDER=YES gen_db
# Iniciar el servicio tg2sip
systemctl start tg2sip.service
###########################################
# Configuración Asterisk
###########################################
# Asterisk 1.6
[telegram]
deny=0.0.0.0/0.0.0.0
type=peer
qualify=yes
permit=10.15.200.13/255.255.255.255
host=tg2sip.masnet.ec
port=5060
fromdomain=tg2sip.masnet.ec
nat=no
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=slin48,opus,slin,slin12,slin24,slin32,slin44,slin96,slin192
# Asterisk 22.2 (Freepbx 17) side:
# /etc/asterisk/pjsip.endpoint.conf
[+5939XXXXXXX]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=slin48,opus,slin,slin12,slin24,slin32,slin44,slin96,slin192
aors=+5939XXXXXXX
send_connected_line=no
rtp_keepalive=0
language=es_419
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
direct_media=no
rtp_symmetric=yes
dtmf_mode=auto
# /etc/asterisk/pjsip.aor.conf
[+5939XXXXXXX]
type=aor
qualify_frequency=60
contact=sip:127.0.0.1:5070
# /etc/asterisk/pjsip.identify.conf
[+5939XXXXXXX]
type=identify
endpoint=+5939XXXXXXX
match=127.0.0.1
# Agregar en /etc/asterisk/codecs.conf
[opus]
type=opus
fec=yes
packet_loss=10
dtx=yes
cbr=yes
bitrate=48000
complexity=8
max_playback_rate=48000
# CallerID desde Telegram:
# /etc/asterisk/extensions_custom.conf
[from-telegram]
exten => _.,1,Set(CALLERID(num)=+${PJSIP_HEADER(read,X-TG-Phone)})
same => n,ExecIf($ ["${CALLERID(num)}"="+"]?Set(CALLERID(num)=${PJSIP_HEADER(read,X-TG-Phone)}))
same => n,Set(CALLERID(name)=${PJSIP_HEADER(read,X-TG-FirstName)} ${PJSIP_HEADER(read,X-TG-LastName)})
include => from-pstn
# Reiniciar asterisk dialplan
asterisk -rx "dialplan reload"
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