Certified integration of Reliance Jio SIP trunks with Asterisk, FreePBX, and ViciDial. Call +91 75999-67999.
Are you planning to configure or troubleshoot a Reliance Jio SIP Trunk? Call Soft Corporation provides expert telecom integration services across India. We specialize in setting up and configuring Jio business voice services with Asterisk, FreePBX, ViciDial, Issabel, and Grandstream UCM IP PBX systems.
Telecom carriers in India like Jio 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 Kanpur.
Here are the standard configuration parameters required for setting up Jio SIP trunks on Asterisk-based PBX servers:
| Parameter | Value / Details |
|---|---|
| Authentication Type | IP-Based or IMS Registration (using +91XXXXXXXXXX@ims.jio.com) |
| VLAN ID | VLAN 880 (default, or custom as per PO) |
| SIP Server/Proxy | jioims.com or specific SBC IP provided |
| Signaling Port | 5060 (UDP/TCP) |
| DTMF Mode | RFC2833 |
| Codecs Supported | G.711 alaw (PCMA), G.711 ulaw (PCMU), G.729 |
1. Physical/VLAN WAN Setup: Jio SIP trunks are provisioned over Jio Business Fiber (FTTx) using a dedicated VLAN (typically VLAN 880 or custom). Jio uses either IP authentication or IMS authentication (requires registration string). A key requirement is configuring static routes on your PBX server so that traffic for Jio's signaling/media networks is routed through the Jio gateway rather than your default internet connection. Disable SIP ALG on your router to avoid registration and one-way audio issues.
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:
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.
One-way audio is usually due to missing static routes on your Asterisk/FreePBX server or router. The RTP media packets must be explicitly routed via Jio's gateway IP instead of the main Internet interface.
Yes, it is highly recommended to use a second Network Interface Card (NIC) on your server specifically dedicated to the Jio Fiber connection to isolate the SIP trunk traffic.
In FreePBX, we create a new Chan_SIP or PJSIP trunk, set the SIP server, configuration headers, and add custom routing tables on the server OS. Contact +91 75999-67999 for remote integration.
We configure and troubleshoot SIP trunks for all major telecom service providers in India:
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