How do I configure a Airtel SIP Trunk on Asterisk?

To configure Bharti Airtel SIP Trunk on Asterisk, you need to create a SIP trunk entry, input the carrier PEER parameters (such as the SBC IP or domain, codecs, DTMF mode), define OS static routes for the dedicated telecom gateway, and set up your outbound dialing rules.

What causes one-way audio on a Airtel SIP trunk in Asterisk?

One-way audio is typically caused by missing static routes on your Linux OS, incorrect NAT settings, or SIP ALG being active on your router. Ensure voice media packets are routed through the telecom interface.

How to Configure Airtel SIP Trunk on Asterisk in Surat

Integrating Airtel SIP Trunk with Asterisk

Integrating a Bharti Airtel SIP Trunk with a Asterisk environment requires specific configuration parameters to ensure outbound call authentication, correct inbound DID routing, and clean audio delivery. Our telecom engineers at Call Soft Corporation have prepared this complete configuration guide to assist call center administrators and network engineers in Surat.

Depending on whether your Airtel trunk uses IP authentication (matching your static IP gateway) or user registration (auth username and secret), you will need to add custom PEER settings and Dialplan routing rules in Asterisk.

1. Asterisk Configuration Settings

Open your Asterisk config files (typically located in `/etc/asterisk/`) and add the following parameters:

                ; 1. Add to /etc/asterisk/sip.conf

                [airtel-sip-trunk]

                type=peer

                host=ims.airtel.in

                context=from-trunk

                disallow=all

                allow=alaw,ulaw

                dtmfmode=rfc2833

                qualify=yes

                nat=force_rport,comedia

                username=+91XXXXXXXXXX@domain
secret=your_password
fromdomain=ims.airtel.in

                ; 2. Add Dialplan rules to /etc/asterisk/extensions.conf

                [outbound-context]

                exten => _9X.,1,Dial(SIP/airtel-sip-trunk/${EXTEN:1})

                exten => _9X.,2,Hangup()

2. Gateway Routing & Firewall Setup

For most Indian operators, voice traffic must be routed via a dedicated gateway interface on your server or router. Ensure that your server is configured with the correct static routes. For example:

              ip route add <SIP_Proxy_IP_Range> via <Telecom_Router_IP>
            

Additionally, disable SIP ALG on your office router or firewall. SIP ALG commonly alters headers and breaks registration or causes one-way audio issues. Ensure UDP ports 5060 (SIP signaling) and UDP ports 10000–20000 (RTP media) are open and forwarded correctly.

Frequently Asked Questions

❓ Why am I getting "403 Forbidden" errors on outbound calls?

This usually occurs if the outbound Caller ID set in your dialer/PBX does not match the active DID or pilot number authorized by Airtel. Double check that the Caller ID format conforms to your telecom provider's requirements.

❓ How do I handle one-way audio on Asterisk?

One-way audio is almost always caused by missing static routes on your Asterisk OS or firewall issues (NAT configuration). Make sure the media IP range of Airtel is routed correctly and RTP port forwarding is enabled.

❓ Can Call Soft Corporation help us configure our dialer?

Yes, we provide complete remote and on-site integration support for all dialers and operators across India. Call our certified engineers at +91 75999-67999 for instant help.

Configure Airtel SIP on Other Platforms

Configure on ViciDial Configure on GoAutoDial Configure on FreePBX Configure on Issabel

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