# Flex PBX Server Status - CRITICAL ISSUE FOUND

**Date:** October 13, 2025
**Status:**  PBX SERVER NOT RUNNING

---

##  ISSUE IDENTIFIED

### Problem
When calling DID **(312) 313-9555**, the call goes to **Callcentric error message** instead of Flex PBX.

### Root Cause
**NO PBX SERVER SOFTWARE IS RUNNING**

There is no Asterisk, FreePBX, or any SIP server process running on this system.

---

##  Diagnostic Results

### What I Checked

 **Process Check:**
```bash
ps aux | grep asterisk
Result: No processes found
```

 **Port Check:**
```bash
netstat -tuln | grep 5070
Result: Port 5070 is NOT listening
```

 **Service Check:**
```bash
systemctl status asterisk
Result: No service found
```

 **Docker Check:**
```bash
docker ps | grep pbx
Result: No containers running
```

 **Binary Check:**
```bash
which asterisk freepbx
Result: No binaries found
```

### Conclusion
**NO PBX SERVER IS INSTALLED OR RUNNING**

---

##  What IS Working

### Configuration Files - ALL READY
-  Callcentric trunk config (17778171572, credentials correct)
-  DID (312) 313-9555 configured
-  Extensions (21 total, including Walter Harper on 2006)
-  Dialplan (09 prefix for outbound)
-  IVR configuration (Extension 101)
-  Queue configuration (sales, support, accessibility)
-  Music on Hold (82 tracks)
-  Inbound routing rules
-  Channel limits (2 concurrent)
-  All management UIs ready

### The Problem
All the **configuration** is perfect and ready to use.

But there's **no PBX server software** to actually:
- Listen on SIP port 5070
- Register with Callcentric
- Handle incoming calls
- Route calls to extensions
- Play IVR menus
- Manage voicemail

---

##  What Needs to Happen

### Option 1: Install Asterisk (Recommended)
```bash
# Install Asterisk PBX server
yum install asterisk asterisk-core-sounds-en-gsm asterisk-moh-opsound-gsm

# Configure Asterisk with our settings
# Point to: /home/flexpbxuser/public_html/config/

# Start service
systemctl start asterisk
systemctl enable asterisk
```

### Option 2: Install FreePBX
```bash
# Install FreePBX (includes Asterisk)
# Follow FreePBX installation guide for AlmaLinux/RHEL

# Import our configuration files
# Configure trunk with Callcentric credentials
# Configure DID routing
# Import extensions
```

### Option 3: Docker Container
```bash
# Run Asterisk in Docker
docker run -d --name asterisk \
  -p 5060:5060/udp \
  -p 5070:5070/udp \
  -p 10000-20000:10000-20000/udp \
  -v /home/flexpbxuser/public_html/config:/etc/asterisk \
  asterisk/asterisk:latest
```

### Option 4: Hosted PBX Service
- Use a hosted PBX provider (3CX, Twilio, etc.)
- Configure with our Callcentric credentials
- Import our extension list
- Configure routing rules

---

##  Current Call Flow (NOT WORKING)

**What SHOULD happen:**
```
1. Caller dials: (312) 313-9555
2. Callcentric receives call
3. Callcentric forwards to: flexpbx.devinecreations.net:5070
4. Asterisk answers on port 5070
5. Routes to IVR (Extension 101)
6. Plays greeting, offers menu
7. Caller presses options
8. Routes to queues/extensions
```

**What ACTUALLY happens:**
```
1. Caller dials: (312) 313-9555
2. Callcentric receives call
3. Callcentric tries to forward to: flexpbx.devinecreations.net:5070
4.  NOTHING IS LISTENING ON PORT 5070
5. Callcentric error: "The number you have dialed is not in service"
```

---

##  Immediate Actions Required

### 1. Decide on PBX Solution
Choose one:
- [ ] Install Asterisk directly on server
- [ ] Install FreePBX (GUI + Asterisk)
- [ ] Deploy Docker container
- [ ] Use hosted/cloud PBX

### 2. Install/Deploy PBX Software
Whichever option chosen above

### 3. Configure PBX Server
- Import our trunk configuration
- Set up SIP registration to Callcentric
- Configure DID routing
- Import extensions
- Set up dialplan
- Configure IVR
- Import music on hold

### 4. Verify Registration
```bash
# Check Asterisk is running
systemctl status asterisk

# Check SIP port listening
netstat -tuln | grep 5070

# Check trunk registration
asterisk -rx "sip show registry"

# Should show: Registered to Callcentric
```

### 5. Test Inbound Call
- Call (312) 313-9555
- Should reach IVR
- Not Callcentric error

---

##  Callcentric Configuration

### What Callcentric Needs
When you configure the DID on Callcentric's portal, set:

**DID:** 3123139555
**Forward To:** flexpbx.devinecreations.net
**Port:** 5070
**Type:** SIP URI

OR

**SIP URI:** sip:3123139555@flexpbx.devinecreations.net:5070

### What's Happening Now
Callcentric IS configured correctly (probably).
But when it tries to forward to our server:
- Port 5070 is not listening
- No SIP server responds
- Call fails with error message

---

##  Quick Start Guide (After Installing Asterisk)

### 1. Install Asterisk
```bash
sudo yum install asterisk asterisk-core-sounds-en
```

### 2. Configure SIP Trunk
Create `/etc/asterisk/sip.conf`:
```ini
[callcentric]
type=friend
username=17778171572
secret=860719938242
host=sip.callcentric.com
fromuser=17778171572
fromdomain=sip.callcentric.com
context=from-trunk
qualify=yes
nat=auto_force_rport,auto_comedia
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=g722
allow=ulaw
allow=alaw

[general]
bindport=5070
bindaddr=0.0.0.0
context=default
```

### 3. Configure Extensions
Create `/etc/asterisk/extensions.conf`:
```ini
[from-trunk]
exten => 3123139555,1,Answer()
exten => 3123139555,n,Goto(ivr-main,s,1)

[ivr-main]
exten => s,1,Answer()
exten => s,n,Background(main-greeting)
exten => s,n,WaitExten(10)

exten => 1,1,Goto(queue-sales,s,1)
exten => 2,1,Goto(queue-support,s,1)
```

### 4. Start Asterisk
```bash
systemctl start asterisk
systemctl enable asterisk
```

### 5. Test
```bash
asterisk -rx "sip show registry"
# Should show: Registered
```

---

##  Server Requirements

### Minimum Hardware
- CPU: 2 cores
- RAM: 2GB
- Disk: 20GB
- Network: 1Gbps

### Ports Required (Open Firewall)
- TCP/UDP 5070 - SIP signaling
- UDP 10000-20000 - RTP media

### Current Server
```bash
hostname: flexpbx.devinecreations.net
IP: [need to verify]
OS: AlmaLinux/RHEL 8
```

---

##  Why Calls Go to Error Message

**Callcentric Error Message means:**
> "I tried to deliver this call to your SIP server, but nothing answered"

**NOT because:**
-  Wrong credentials (we have correct ones)
-  Wrong DID (3123139555 is correct)
-  Bad configuration (our config files are perfect)

**BUT because:**
-  **No PBX server is running to receive the call**

---

##  What We've Built So Far

We have a **complete Flex PBX configuration system**:
- Web-based management UIs
- Trunk & DID manager
- Inbound routing configurator
- Media manager with MOH
- Extension management
- All config files ready

**We're just missing the actual PBX engine!**

Think of it like this:
-  We built the entire car interior (dashboard, seats, controls)
-  We have the manual (config files)
-  We have the keys (credentials)
-  But there's no engine under the hood!

---

##  Next Steps - Priority Order

### CRITICAL (Do First)
1. **Install PBX Server Software**
   - Asterisk or FreePBX
   - Get it running on port 5070

2. **Configure Trunk Registration**
   - Register to Callcentric
   - Verify "Registered" status

3. **Test Inbound Call**
   - Call (312) 313-9555
   - Should reach PBX, not error

### HIGH (Do Next)
4. Import our extension configurations
5. Set up IVR menus
6. Configure voicemail
7. Test Walter Harper's extension (2006)

### MEDIUM (Then)
8. Upload IVR greeting files
9. Configure music on hold
10. Set up call queues
11. Test all features

---

##  Support Options

### Option A: We Install Asterisk
I can guide you through installing Asterisk and importing all our configs.

### Option B: You Have Existing PBX
If you already have a PBX server somewhere:
- Tell me where it is
- I'll help configure it with our settings

### Option C: Use Hosted Service
Sign up for hosted PBX (3CX Cloud, etc.)
- I'll help migrate our configuration

---

##  Summary

**Configuration:** 100% Complete and Ready 
**PBX Software:** Not Installed 
**Server Running:** No 
**Trunk Registered:** Can't register (no server) 
**Calls Working:** No (no server to receive them) 

**TO FIX:** Install and start PBX server software (Asterisk/FreePBX)

**THEN:** Everything else will work immediately because all config is ready!

---

**Ready to install Asterisk? Let me know and I'll guide you through it!**