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Airtel SIP Trunk Configuration in Bhubaneswar

Certified integration of Bharti Airtel SIP trunks with Asterisk, FreePBX, and ViciDial. Call +91 75999-67999.

Airtel SIP Trunk Integration & Setup

Are you planning to configure or troubleshoot a Bharti Airtel SIP Trunk? Call Soft Corporation provides expert telecom integration services across India. We specialize in setting up and configuring Airtel business voice services with Asterisk, FreePBX, ViciDial, Issabel, and Grandstream UCM IP PBX systems.

Telecom carriers in India like Airtel deliver SIP trunks over dedicated physical fiber Leased Lines (ILL), MPLS, or FTTX connections. Proper configuration requires precise settings for WAN interfaces, IP-based or registration-based authentication, custom static routing, and dialing rules. Our certified engineers ensure 100% reliable connectivity with zero call drops, high call quality (G.711 alaw), and proper DID mapping for your business in Bhubaneswar.

Technical Specifications (Airtel)

Here are the standard configuration parameters required for setting up Airtel SIP trunks on Asterisk-based PBX servers:

Parameter Value / Details
Authentication Type Register String / Username & Password or IP-Authenticated
SIP Domain / Host ims.airtel.in or carrier-provided IP
Signaling Port 5060 (UDP/TCP)
From Domain ims.airtel.in
DTMF Mode RFC2833 or SIP INFO
Codecs Supported G.711 alaw, G.729, G.722

Step-by-Step Configuration Guide

1. Physical/VLAN WAN Setup: Airtel SIP trunks are delivered over dedicated Leased Lines (MPLS) or standard Internet connections. They commonly use the domain `ims.airtel.in` or specific SBC IPs. For IMS-based setups, you must use a registration string with your pilot number in the format `+91XXXXXXXXXX@ims.airtel.in`. A critical setting is specifying the correct outbound Caller ID (which must match your active DID ranges) and setting `fromdomain=ims.airtel.in` in your SIP trunk settings.

; Sample Peer Details for Asterisk (sip.conf)
[airtel-trunk]
type=peer
host=ims.airtel.in
context=from-trunk
disallow=all
allow=alaw
dtmfmode=rfc2833
qualify=yes
nat=force_rport,comedia
username=+91XXXXXXXXXX@domain
secret=your_password

2. Define OS Level Static Routes: You must route the SIP signaling IP and RTP voice IP ranges through the telecom gateway gateway IP rather than your public internet link. On Linux, run:

ip route add <SIP_PROXY_IP> via <TELECOM_GATEWAY_IP> dev eth1

3. Outbound Dialplan: In FreePBX or ViciDial, define the dialing prefix and outbound Caller ID to match your allocated pilot number exactly. Contact our team at +91 75999-67999 for custom scripting or troubleshooting help.

Frequently Asked Questions

❓ How do I register an Airtel SIP trunk in Asterisk?

You must configure a registration string under your trunk settings: `username:password@ims.airtel.in/pilot_number`. Ensure DNS is set up to resolve `ims.airtel.in` through Airtel's DNS servers.

❓ What happens if outbound Caller ID is set incorrectly on Airtel SIP Trunk?

Airtel SIP trunks will reject outbound calls (usually with a 403 Forbidden or 503 Service Unavailable error) if the outgoing Caller ID does not belong to your allocated DID range.

❓ Can we route incoming calls to specific DIDs?

Yes, Airtel provides a range of DIDs. We can map these DIDs in your FreePBX/Asterisk Inbound Routes to direct calls to the right extensions or IVR systems.

Other Telecom Operators

We configure and troubleshoot SIP trunks for all major telecom service providers in India:

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