How do I configure a Reliance SIP Trunk on ViciDial?

To configure Reliance Communications SIP Trunk on ViciDial, 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 Reliance SIP trunk in ViciDial?

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 Reliance SIP Trunk on ViciDial

Integrating Reliance SIP Trunk with ViciDial

Integrating a Reliance Communications SIP Trunk with a ViciDial 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.

Depending on whether your Reliance 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 ViciDial.

1. ViciDial Configuration Settings

In the ViciDial Admin interface, go to Admin -> Carriers -> Add New Carrier and set:

                Carrier ID: RELIANCESIP

                Carrier Name: Reliance SIP Trunk

                Registration String: ; Not required for IP-based trunks

                Account Entry:

                [reliance_vicidial]

                type=peer

                host=CARRIER_SBC_IP

                context=trunkinbound

                disallow=all

                allow=alaw

                dtmfmode=rfc2833

                qualify=yes

                nat=force_rport,comedia

                insecure=port,invite

                Protocol: SIP

                Globals String: RELIANCEDIAL = SIP/reliance_vicidial

                Dialplan Entry:

                exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)

                exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@reliance_vicidial,,tTo)

                exten => _91NXXNXXXXXX,3,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 Reliance. Double check that the Caller ID format conforms to your telecom provider's requirements.

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

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 Reliance 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 Reliance SIP on Other Platforms

Configure on GoAutoDial Configure on FreePBX Configure on Issabel Configure on Asterisk

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