(D241024) IP PBX call distributor VoIP Gateway GSM/3G/4G/5G for home and small office

PROOF OF CONCEPT

1. Requirement and concept

We're building a PBX system for our business. It will handle:

• automatic distribute incoming calls
• call outside by specific phone numbers, automatic route to reduce call cost
• sms send api for OTP maybe?

Since we don't have any SIP trunk from our carriers, we will take incoming calls from regular sim cards.

• Disadvantage:
  • Single-line: only one concurrent call per SIM card at a time
  • Need more work to handle system failure due to OS, power..
• Not sure:
  • Voice over LTE (VoLTE) are supported by all carriers.
  • Voice over Wi‑Fi (VoWiFi) are supported by VinaPhone, Mobifone?

Call flow as bellow:

Our stack:

• Hardwares:
  • Mini PC / Router: ZOTAC ZBOX CI323 nano
  • 2G/3G/4G/5G module: Quectel EC20CEFRG-MINIPCIE-C (Vietnam frequency bands: B1/B3/B8), PCIE to USB adapter (sim card slot included), antena
  • VoIP phone: haven't bought yet
• Softwares
  • OS: TrueNAS SCALE 24.10 - Electric Eel
  • VM: jailmaker
  • PBX: Asterisk
  • Softphone: Linphone (Linux / iOS), Baresip (Android), MicroSIP (Windows)

2. Step by step

2.1 Asterisk Installation

• OS: Debian 12 ("Bookworm")

$ mkdir tmp
$ cd tmp/
$ wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22-current.tar.gz
$ tar zxvf asterisk-22-current.tar.gz
$ cd asterisk-22*/
$ sudo ./contrib/scripts/install_prereq install
$ ./configure
$ make menuselect

Most of IP phones are support G722 and / or G729 codecs now.

• Under Add-ons:
  • Select [*] format_mp3
• Under Codec Translators:
  • Select [*] codec_opus
  • Select [*] codec_silk
  • Select [*] codec_siren7
  • Select [*] codec_siren14
  • Select [*] codec_g729a
• Under Core Sound Packages:
  • Deselect [*] CORE-SOUNDS-EN-GSM
  • Select [*] CORE-SOUNDS-EN-WAV
  • Select [*] CORE-SOUNDS-EN-G722
  • Select [*] CORE-SOUNDS-EN-G729
• Under Extras Sound Packages:
  • Select [*] EXTRA-SOUNDS-EN-WAV
  • Select [*] EXTRA-SOUNDS-EN-G722
  • Select [*] EXTRA-SOUNDS-EN-G729

$ make
$ sudo ./contrib/scripts/get_mp3_source.sh
$ sudo make install
$ sudo make config
$ sudo make samples
$ sudo mkdir /etc/asterisk/samples
$ sudo mv /etc/asterisk/*.*  /etc/asterisk/samples/
$ sudo asterisk -rvvvv

Most important config files:

• extensions.conf
• pjsip.conf

2.2. Virtual machine vm1

Purpose: VoIP GSM Gateway to handle voice call from and or to 3G/4G/5G MNOs.

• OS: Debian 12
• IP: 192.168.1.11
• Software: Asterisk
• Driver: asterisk-chan-quectel

2.3. Virtual machine vm0

Purpose: Local PBX.

• OS: Debian 12
• IP: 192.168.1.10
• Software: MikoPBX

3. Resource

Softwares:

• Servers:
  • Asterisk
    • asterisk-chan-quectel
    • asterisk-chan-dongle
  • FreeSWITCH
  • Kamailio
  • Yate
  • Routr
  • FreePBX
  • MikoPBX
  • docker-asterisk
  • asterisk-sound-generator
  • Light Speed for text-to-speech
• Clients:
  • Baresip
  • Linphone
  • Jami
  • MicroSIP
• Docs:
  • Official Asterisk documentation
  • Official Asterisk YouTube Channel
  • Asterisk: The Definitive Guide
• Ref:
  • 使用EC20模块配合asterisk及freepbx实现短信转发和网络电话
  • 如何把SIM卡从手机中取出
  • Quectel LTE 모뎀으로 Asterisk에서 KT, SKT, LG U+ VoLTE 전화 사용하기
  • Simple test bed set-up
  • Thực trạng mạng di động viễn thông tại Việt Nam

4. Miscellaneous

• Linux KVM is much more complicated than FreeBSD Bhyve so maybe we sould run FreeBSD host and Linux guest
• Debian has no Asterisk in their repo, we moved our guest to Ubuntu since we don't want to complie Asterisk ourself
• Run MikoPBX as a Local PBX, Asterisk as a trunk provider. We found that MikoPBX is better than FreePBX in our case.
• Call quality seems not good enough, need to check jitter buffer again
• https://github.com/RoEdAl/asterisk-chan-quectel

test@vm0:~$ cat /etc/asterisk/pjsip.conf 
;===============TRANSPORT

[system-udp]
type=transport
protocol=udp
bind=0.0.0.0

;===============TRUNK

[miko0]
type=aor
contact=sip:192.168.1.11:5060

[miko0]
type=endpoint
context=quectel-incoming
disallow=all
allow=ulaw
aors=miko0

[miko0]
type=identify
endpoint=miko0
match=192.168.1.11

test@vm0:~$ cat /etc/asterisk/extensions.conf 
[quectel-incoming]
exten => _+84XXXXXXXXX,1,Dial(PJSIP/200@miko0)
exten => s,n,Hangup()

edit /etc/asterisk/indications.conf on vm1

#!/bin/bash 

configPath="$1" # Path to the original config file

sed -i 's/country=ru/country=vn/g' "$configPath"
cat >> "$configPath" <<EOF

[vn]
description = Vietnam
; Clone from China
ringcadence = 1000,4000
dial = 450
busy = 450/350,0/350
ring = 450/1000,0/4000
congestion = 450/700,0/700
callwaiting = 450/400,0/4000
dialrecall = 450
record = 950/400,0/10000
info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
stutter = 450+425
EOF

5. Test

Number: 0918026858

• [x] Voice
• [ ] SMS
• [ ] 4G internet?

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