SIP-GSM Gateway

Android app that bridges GSM calls (local SIM) with the CallAgent SIP/Asterisk server.

Dialer	SIP Registration

How It Works

A dedicated rooted Android phone with a local SIM card acts as a SIP-to-GSM gateway:

• Inbound: Someone calls the local number → phone auto-answers → bridges to Asterisk via SIP → AI agent handles the call
• Outbound: Asterisk sends SIP INVITE with X-GSM-Forward: +<number> header → phone dials the destination via GSM SIM → bridges audio back to SIP

Audio flows through shared speaker/mic — both GSM and SIP audio run concurrently on the same hardware, enabled by a Magisk module that disables Android's audio concurrency restrictions.

Audio Codec

G.722 wideband (16 kHz, 64 kbps) for high quality voice. Falls back to G.711 A-law if needed.

Requirements

• Device: Samsung Galaxy S10e (or similar) with LineageOS + Magisk root
• SIM: SIM card with voice plan
• Network: Stable WiFi connection
• Power: Always connected to charger
• Build host: Linux with JDK 17+

Build

chmod +x build.sh
./build.sh          # debug build
./build.sh release  # release build

Outputs:

• gateway-magisk.zip — Magisk module containing the APK, permissions, and audio tools (tinymix, tinycap). This is the only file you need to install.

Device Setup

Only the Magisk module needs to be installed — it includes the APK and handles all permissions automatically.

1. Install Magisk module: Copy gateway-magisk.zip to device, install via Magisk Manager → Modules
2. Reboot the device — the module installs the APK as a privileged system app and grants all permissions on boot
3. Set as default phone app: Settings → Apps → Default apps → Phone app → SIP-GSM Gateway
4. Configure SIP: Enter your Asterisk server address, port, username, and password
5. Start: Tap START — the app registers with Asterisk and begins bridging calls

Asterisk Configuration

1. Create a SIP account for the gateway

Add to sip.conf or create via the realtime database:

[gateway-gw1](agent-template)
secret = <strong-password>
context = gateway-incoming

2. Add gateway dialplan context

Add to extensions.conf:

; Gateway incoming calls (GSM → SIP → Agent)
[gateway-incoming]
exten => _X.,1,NoOp(Gateway call from ${CALLERID(num)} via GSM SIM)
same => n,Set(CDR(destination)=${EXTEN})
same => n,Set(CDR(userfield)=gateway-gw1)
; Route to AI agent (same logic as incoming-calls)
same => n,Set(AgentToUse=${ODBC_AGENT_LOOKUP(gateway-gw1)})
same => n,GotoIf($["${AgentToUse}" = ""]?default_agent:route_to_agent)
same => n(route_to_agent),MixMonitor(/var/spool/asterisk/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.wav)
same => n,Dial(SIP/${AgentToUse},60,tT)
same => n,Hangup()
same => n(default_agent),MixMonitor(/var/spool/asterisk/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.wav)
same => n,Dial(SIP/100,60,tT)
same => n,Hangup()

; Outbound: Agent calls a number via the gateway
; The agent context already allows outbound calls:
;   Dial(SIP/gateway-gw1,,X-GSM-Forward: +1234567890)
; Or use a custom AGI/ARI to set the header.

3. Making outbound calls through the gateway

From Asterisk dialplan, to call a number via the gateway:

exten => _X.,1,NoOp(Outbound via GSM gateway: ${EXTEN})
same => n,SIPAddHeader(X-GSM-Forward: +${EXTEN})
same => n,Dial(SIP/gateway-gw1,60)
same => n,Hangup()

Architecture

┌─────────────────┐     GSM      ┌──────────────────┐
│  Remote Caller   │◄───────────►│  Android Phone    │
│  (local #)       │   voice     │  (S10e + SIM)     │
└─────────────────┘              │                    │
                                 │  ┌──────────────┐ │
                                 │  │ InCallService │ │  GSM call control
                                 │  └──────┬───────┘ │
                                 │         │         │
                                 │  ┌──────▼───────┐ │
                                 │  │ Orchestrator  │ │  Bridges GSM ↔ SIP
                                 │  └──────┬───────┘ │
                                 │         │         │
                                 │  ┌──────▼───────┐ │
                                 │  │  SIP Client   │ │  Registration + calls
                                 │  │  RTP Session  │ │  G.722 audio stream
                                 │  └──────┬───────┘ │
                                 └─────────┼─────────┘
                                           │ SIP/RTP
                                           │ (WiFi)
                                 ┌─────────▼─────────┐
                                 │  Asterisk Server   │
                                 │  (CallAgent SIP)   │
                                 └─────────┬─────────┘
                                           │
                                 ┌─────────▼─────────┐
                                 │  AI Voice Agent    │
                                 └───────────────────┘

Magisk Module

The gateway-magisk.zip module does two critical things:

1. Disables audio concurrency restrictions (system.prop):

   • voice.voip.conc.disabled=false — allows VoIP audio during GSM calls
   • voice.record.conc.disabled=false — allows audio recording during calls
   • voice.playback.conc.disabled=false — allows audio playback during calls

2. Grants system-level permissions (privapp-permissions-gateway.xml):

   • CAPTURE_AUDIO_OUTPUT — capture audio from other sources
   • MODIFY_PHONE_STATE — control telephony
   • READ_PRECISE_PHONE_STATE — detailed call state info

Troubleshooting

• One-way audio: Ensure the Magisk module is installed and device is rebooted
• Echo: The app uses Android's AcousticEchoCanceler + VOICE_COMMUNICATION mode
• SIP not registering: Check WiFi connectivity, server address, and credentials
• Calls not auto-answering: Ensure the app is set as the default phone app
• Audio drops: Check WiFi stability; the app holds a WiFi lock but poor signal will cause issues