Airtel Telecommunication
FreePBX / Vicidial
How to Configure Airtel SIP Trunking & PRI Line on Vicidial and FreePBX
Telecom Engineering Team
August 12, 2026
8 Min Read
Bharti Airtel provides enterprise SIP Trunking over dedicated copper or fiber MPLS links using IP-based Authentication (without username/password). Here is the verified configuration for call centers running Vicidial, GoAutoDial, or FreePBX.
Prerequisites from Airtel Technical Team
- Airtel SBC IP: e.g.,
10.232.xxx.xxx(Signaling Gateway) - Media IP Subnet: Allocated range for RTP voice streams
- DDI Pilot Range: Assigned 10-digit DDI numbers (e.g., 011-40xxxxxx)
- Codecs Allowed: G.711 alaw, G.711 ulaw, G.729
1. Vicidial Carrier Configuration for Airtel
In Vicidial Admin Panel → Carriers → Add A New Carrier, enter the following Account Entry and Dialplan Entry:
; === Vicidial Carrier Account Entry for Airtel SIP Trunk ===
[airtel_sip_out]
disallow=all
allow=alaw
allow=ulaw
allow=g729
type=peer
host=10.232.14.10 ; Replace with your Airtel SBC IP
port=5060
insecure=port,invite
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context=trunkinbound
Vicidial Dialplan Entry (Outbound Prefix 91):
exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,2,Dial(SIP/airtel_sip_out/\,60,tTor)
exten => _91X.,3,Hangup
2. FreePBX PJSIP Trunk Settings for Airtel
In FreePBX GUI → Connectivity → Trunks → Add PJSIP Trunk:
- Trunk Name:
Airtel_SIP_Trunk - Authentication:
None(IP Authenticated) - SIP Server:
10.232.14.10 - Context:
from-pstn - Direct Media:
No(Prevents One-Way Audio)
Facing One-Way Audio on Airtel?
Ensure that your local firewall (iptables/firewalld) permits UDP ports 10000-20000 for RTP. Need engineer support? Call +91 75999-67999 for immediate remote setup.