How to Configure BSNL SIP Trunk with FreePBX in Pune
Overview & Network Topology
Setting up high-performance voice solutions in **Pune** requires proper static IP routing, session settings, and correct trunk peer configurations. For call centers, IT offices, and enterprise networks in Pune utilizing **FreePBX Administration GUI**, integrating a **BSNL Enterprise NGN SIP Trunk** provides crystal-clear calling and massive savings over traditional PRI cards.
Below is the step-by-step configuration layout to connect your FreePBX system to the BSNL Session Border Controller (SBC).
Step 1: Network Routing & Gateways
BSNL SIP trunks do not run over standard public gateways. They require static routing rules on your Linux server (CentOS/Ubuntu) or core firewall (Sophos, Mikrotik) so that SIP (5060) and RTP (10000-20000) packets reach their voice softswitch.
Run this route command on your FreePBX shell:
*Replace <Your_Operator_Gateway_IP> with the WAN gateway IP provided in your BSNL dossier.
Step 2: FreePBX Trunk Configurations
Go to Connectivity > Trunks in your control panel and input the following configuration parameters:
Step 3: Dial Patterns & Outbound Routes
Ensure your outbound caller ID matches the Pilot DID number provided by BSNL. Outbound calls dialing pattern should strip prefixes properly. For instance, if you dial 9 before dialing a mobile number, configure the route pattern to strip the first digit and send the remaining digits to the trunk.
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DID Mapping
Map your DID range (inbound calls) to an IVR, Queue, or specific Extension in your Inbound Routes.
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Outbound CLI Matching
Telecom operators in India reject calls if Caller Line Identification (CLI) does not match your allotted pool.
Looking for expert PBX configuration in Pune?
Setting up Session Border Controllers (SBC), resolving NAT one-way audio issues, and dealing with telecom routing tables can be complicated. We offer full remote and on-site engineering support for call centers, offices, and hotels in **Pune**.
Configuration Troubleshooting FAQs (Pune)
1. Why am I getting "403 Forbidden" or "All Circuits Busy" on outbound calls?
This usually occurs when the outbound Caller ID (CLI) configured on your FreePBX trunk does not exactly match the pilot number or DID ranges allotted to your trunk by BSNL.
2. How to fix one-way audio (mute calls) after configuration?
One-way audio is typically caused by NAT or firewall blockages. Ensure UDP ports 10000-20000 are open in your router and that the Local Networks/External Address parameters are correctly set in your Asterisk SIP settings.
3. Can you do this setup remotely for our office in Pune?
Yes! We provide complete Remote Configuration, Firewall Audits, and AMC support across Pune. Contact our team at +91 75999-67999 for quick pricing and availability.