This guide explains how to connect FreeSWITCH → Asterisk (PRI server) and route calls (e.g., numbers starting with 5) from FreeSWITCH to Asterisk.

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🧭 Architecture Overview

SIP Phone (1000 / 1003 / etc)
        ↓
   FreeSWITCH
        ↓ (SIP Trunk)
   Asterisk Server
        ↓
      PRI Line
        ↓
   PSTN (Mobile Number)

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🎯 Use Case

• Users dial numbers like: 578776152252
• FreeSWITCH:
  • Removes 5
  • Sends 078776152252to Asterisk
• Asterisk:
  • Routes call via PRI

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⚙️ 1. Asterisk Configuration

📍 File:

/etc/asterisk/sip.conf

✅ SIP Trunk (IP-based authentication)

[from_10]
type=peer
host=192.168.200.10
context=tata
insecure=invite,port
disallow=all
allow=ulaw,alaw
qualify=yes
nat=no

🔍 Key Points

Parameter	Description
host	FreeSWITCH IP
context=tata	Incoming calls go to this dialplan
insecure=invite,port	Disable auth (IP-based trust)
type=peer	Best for trunk

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📞 2. Asterisk Dialplan

📍 File:

/etc/asterisk/extensions.conf

✅ Example:

[tata]exten => _0X.,1,NoOp(Call from FreeSWITCH)
exten => _0X.,n,Dial(SIP/${EXTEN}@TATA)
exten => _0X.,n,Hangup()

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🔁 Reload Asterisk

asterisk -rx "sip reload"
asterisk -rx "dialplan reload"

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⚙️ 3. FreeSWITCH Gateway Configuration

📍 File:

/usr/local/freeswitch/conf/sip_profiles/external/asterisk_pri.xml

✅ Gateway Config:

<include>
  <gateway name="asterisk_pri">
    <param name="proxy" value="192.168.200.19"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

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🔁 Reload FreeSWITCH

fs_cli -x "reloadxml"
fs_cli -x "sofia profile external restart"

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🔍 Verify Gateway

fs_cli
sofia status gateway asterisk_pri

Expected:

Status: UP

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⚙️ 4. FreeSWITCH Dialplan

📍 File:

/usr/local/freeswitch/conf/dialplan/default.xml

✅ Routing Rule

<extension name="send-to-asterisk">
  <condition field="destination_number" expression="^5(\d+)$">    <!-- Remove first digit (5) -->
    <action application="set" data="destination_number=0${destination_number:1}"/>    <!-- Send call to Asterisk -->
    <action application="bridge" data="sofia/gateway/asterisk_pri/${destination_number}"/>  </condition>
</extension>

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👤 5. FreeSWITCH Users (Extensions)

📍 Directory:

/usr/local/freeswitch/conf/directory/default/

✅ Example (1000)

<include>
  <user id="1000">
    <params>
      <param name="password" value="1000"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_number" value="1000"/>
    </variables>
  </user>
</include>

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🔄 Reload Users

fs_cli -x "reloadxml"

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🚀 Call Flow (Working)

Example:

Extension 1003 dials → 578776152252

Flow:

1003
↓
FreeSWITCH
↓ (dialplan match ^5\d+)
remove 5 → 078776152252
↓
sofia/gateway/asterisk_pri
↓
Asterisk (context=tata)
↓
PRI
↓
Mobile number connected

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❗ Common Issues & Fixes

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🔴 Issue 1: “extension not found in context default”

Cause:
Asterisk not matching SIP peer.

Fix:

• Use type=peer
• Ensure host=FreeSWITCH_IP
• Use insecure=invite,port

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🔴 Issue 2: “username mismatch”

username mismatch, have <1003>, digest has <FreeSWITCH>

Cause:
Auth conflict between extension and trunk.

Fix:
👉 Use IP-based authentication (no username/password)

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🔴 Issue 3: FreeSWITCH error USER_NOT_REGISTERED

Cause:
Trying to call non-registered SIP user instead of gateway.

Fix:
Use:

sofia/gateway/asterisk_pri/number

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🔴 Issue 4: Gateway NOREG

Normal when:

register=false

✔ Not an issue.

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🔧 Debug Commands

Asterisk

asterisk -rvvv
sip set debug on

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FreeSWITCH

fs_cli
sofia status
sofia status gateway asterisk_pri

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🏁 Final Result

✔ FreeSWITCH extensions (1000–1010 etc.) can dial
✔ Numbers starting with 5 routed to Asterisk
✔ Asterisk sends calls via PRI
✔ No authentication issues
✔ Stable SIP trunk