Enterprise SIP Trunking Service - Connect Your PBX to the Cloud
Talkloo's SIP trunking service connects your IP PBX to the global PSTN over the cloud - elastic channels, tier-1 carrier interconnection, and Pakistan + international DID numbers, all managed from a single dashboard. Unlike every other SIP trunk provider, Talkloo bundles SIP trunking, DID management, and AI voice agents on one platform. No stitching together three separate vendors.
Built for enterprise IT teams, call center managers, and PBX administrators replacing ISDN/PRI lines.
No contracts, no channel fees.
Channels scale automatically - pay only for what you use.
Direct carrier routes, G.711/G.729, low latency.
Automatic re-routing to backup carriers.
From WhatsApp call to your PBX or BPO – with context
The AI voice agent answers first. When a human is needed it warm-transfers over SIP to your PBX, in-house call centre or outsourced BPO with the transcript and CRM record attached.
What Is SIP Trunking?
SIP trunking replaces traditional ISDN/PRI lines with virtual phone connections delivered over IP. Your IP PBX sends SIP INVITE messages to a SIP trunk provider; the provider authenticates the trunk and routes the call to the PSTN. Voice travels as RTP packets over your data network - no copper, no fixed circuits.
The result: elastic capacity that scales with demand, pay-per-use pricing, and global PSTN reach without a single physical line. Any enterprise running Asterisk, FreePBX, Cisco CUCM, 3CX, or any SIP-compliant PBX can migrate from PRI to SIP in hours, not months.
Who needs it: enterprises consolidating carrier contracts, call centers running high-volume inbound/outbound traffic, and businesses that want to keep their existing PBX while moving connectivity to the cloud.
PRI vs SIP Trunking
From fixed circuits to elastic capacity
Fixed channels, physical circuits, months to provision, per-line fees.
Elastic channels, pay-per-use, global reach, live in hours.
Asterisk / CUCM / 3CX
SIP proxy
RTP over IP
Migrate any SIP-compliant PBX from PRI to SIP in a matter of hours.
Everything in Talkloo's SIP Trunking Service
Elastic capacity, tier-1 routes, DIDs, failover, AI, and analytics - a complete enterprise SIP trunking platform in one place.
Elastic SIP Trunks
Channels scale up and down automatically - no fixed channel limits, no pre-provisioning. Traffic spikes during campaigns or peak hours don't drop calls; Talkloo's elastic SIP trunking platform absorbs the burst and scales back when demand drops. You pay only for what you use.
Elastic Capacity
Scales with demand, automatically
Campaign spikes and peak hours absorb into the trunk - no dropped calls, no pre-buying.
Carrier Interconnection
Direct routes, no reseller hops
Tier-1 Carrier Interconnection
Talkloo connects directly to tier-1 carriers - no reseller hops, no quality degradation. Calls traverse the shortest path to the PSTN. Supported codecs: G.711 (ulaw/alaw) and G.729, covering every major PBX and softswitch configuration. Low latency, high MOS scores, consistent call quality on every route.
Global PSTN Connectivity
Terminate and originate calls across multiple countries from a single VoIP SIP trunk. Pakistan coverage is native - Karachi, Lahore, Islamabad, and major cities - plus international routes for enterprises with cross-border operations. One SIP trunk provider for your entire footprint.
Coverage
One trunk, global reach
DID Management
Assign DIDs to trunks in seconds
Karachi landline · bound to SIP trunk
London, UK · local caller ID
Keep your current business numbers - no third-party vendor required.
DID Number Management
Rent and manage landline and mobile DID numbers directly from the Talkloo dashboard. Assign DIDs to SIP trunks in seconds. Port existing numbers. Local caller ID on outbound calls. No third-party number vendor required.
Failover & Redundancy
Automatic failover routing kicks in if a primary route degrades. Calls re-route to backup carriers without manual intervention. Talkloo's hosted SIP trunk infrastructure is built for 99.9% uptime SLA - disaster recovery is built in, not bolted on.
Failover Routing
Built-in disaster recovery
Tier-1 carrier
Auto-detect
No manual steps
SIP Trunk + AI
Trunk to AI, no middleware
SIP trunk
24/7
full context
Bind a trunk directly to an AI voice agent - no middleware, no extra vendor.
SIP Trunk + AI Voice Agent
This is where Talkloo's SIP trunking platform separates from every generic SIP trunk service provider. Bind your SIP trunk directly to a Talkloo AI voice agent - inbound calls are answered by a human-like AI, 24/7, with no extra integration layer. Outbound AI calling campaigns run on the same trunk. SIP trunk voice AI, fully unified.
Call Recording & Analytics
Record every SIP trunk call - dual-channel, on-demand, or always-on. Real-time monitoring dashboard shows active channels, call quality metrics, and live CDR feeds. Fraud detection flags anomalous traffic patterns before they become a problem. Full CDR export for billing reconciliation and compliance.
Monitoring
Full visibility, live CDR
Full CDR export for reconciliation, audit, and regulatory requirements.
SIP Trunking Use Cases
Migration, call center infrastructure, failover, AI calling, and enterprise consolidation - all on one SIP trunking platform.
PBX to Cloud Migration
Replace ISDN/PRI lines with cloud SIP trunks while keeping your existing PBX - Asterisk, FreePBX, Cisco CUCM, 3CX, Avaya, Yeastar. Point your trunk settings at Talkloo's SIP server, assign DIDs, and you're live. No hardware swap, no downtime window measured in weeks.
Call Center SIP Infrastructure
High-volume inbound/outbound SIP trunking for contact centers. Elastic channels handle agent queues and IVR traffic without pre-buying capacity. Pair with Talkloo's cloud VoIP service for a complete call center voice stack - SIP trunking, routing, recording, and analytics in one place.
Backup & Failover Trunking
Run Talkloo as a secondary SIP trunk behind your primary carrier. If the primary fails, calls route through Talkloo automatically. Zero cost unless you use it - the definition of a cost-effective disaster recovery strategy for enterprise telephony.
AI-Powered Calling
Connect SIP trunks to Talkloo's AI-powered calling engine for automated inbound handling and outbound campaigns. Programmable SIP routing lets you define call flows, trigger AI agents on specific DIDs, and hand off to live agents when needed - all without a separate CPaaS vendor.
Enterprise PBX Consolidation
Replace multiple carrier contracts - one per region, one per office - with a single enterprise SIP trunk from Talkloo. Centralized provisioning, unified billing, one support contact. The business SIP trunk provider model that actually simplifies operations.
Why Choose Talkloo as Your SIP Trunk Provider
Compatible PBX Systems
Your existing PBX, connected
TLS + SRTP for encrypted media. IP authentication and credential-based authentication both supported.
Browser-based endpoints with Opus, transcoded to G.711 at the trunk boundary.
Compatible With Your Existing PBX
Talkloo's SIP trunking service works with any SIP-compliant PBX or softswitch. Tested and compatible systems:
Asterisk & FreePBX
Configure a SIP/PJSIP trunk with Talkloo credentials in pjsip.conf or the GUI. FreePBX: add a Chan_PJSIP trunk with the Talkloo server address and credentials.
Cisco CUCM & 3CX
Configure a SIP trunk in CUCM pointing to Talkloo's SIP proxy. For 3CX, add Talkloo as a generic SIP trunk in the Management Console.
Avaya, Yeastar & Grandstream
Configure via Session Manager SIP entity, Yeastar PBX System, or Grandstream UCM VoIP trunk under SIP settings. Any SIP-compliant softswitch or SBC works - standard SIP over UDP/TCP/TLS, port 5060/5061.
This is cloud PBX SIP trunking that meets your infrastructure where it is - not the other way around.
How to Get Started
From signup to live trunk in under 15 minutes on a standard PBX.
Step 1 - Configure Your SIP Credentials
Sign up and open the Talkloo dashboard. Navigate to SIP Trunking → New Trunk. The dashboard generates your SIP username, password, and server address instantly. No ticket, no waiting for a provisioning team.
Step 2 - Point Your PBX to Talkloo
Enter the SIP credentials in your PBX trunk settings. Works with any SIP-compliant system - Asterisk, FreePBX, 3CX, Cisco, Yeastar, Grandstream. Set your outbound route to use the new trunk. Inbound routing is configured in the Talkloo dashboard. The whole process takes under 15 minutes on a standard PBX.
Step 3 - Assign DIDs & Go Live
Rent DID number management numbers from the dashboard - Pakistan local numbers, international numbers, or both. Assign them to your SIP trunk. Run a test call. Go live. That's the entire onboarding flow for a hosted SIP trunk on Talkloo.
Need help with a complex enterprise deployment? Talk to our team - we support Cisco CUCM, Avaya, and multi-site PBX consolidation projects directly.
Frequently asked questions
The most common questions from businesses evaluating an enterprise SIP trunk provider.
What is SIP trunking and how does it work?
SIP trunking replaces physical ISDN/PRI lines with virtual phone connections over IP. Your PBX sends a SIP INVITE to Talkloo's SIP proxy; Talkloo authenticates the trunk and routes the call to the PSTN. Voice travels as RTP packets over your internet connection. The result is elastic capacity - channels scale on demand - with no fixed circuits and no per-channel fees.
Can I connect my existing PBX (Asterisk, 3CX, Cisco) to Talkloo SIP trunking?
Yes. Talkloo's SIP trunking service is compatible with any SIP-compliant PBX or softswitch - Asterisk, FreePBX, 3CX, Cisco CUCM, Avaya, Yeastar, Grandstream, and others. You configure a SIP trunk in your PBX settings using Talkloo's SIP server address and credentials. Standard SIP over UDP/TCP/TLS on port 5060/5061. No proprietary hardware or software required.
Does Talkloo provide DID numbers for Pakistan?
Yes. Talkloo provides local DID numbers for Karachi, Lahore, Islamabad, and other major Pakistani cities, plus international DIDs for cross-border operations. Numbers are rented and managed directly from the Talkloo dashboard - no third-party number vendor. Local caller ID on outbound calls is supported.
What codecs does Talkloo SIP trunking support?
Talkloo supports G.711 ulaw, G.711 alaw, and G.729. G.711 is the default and works across all compatible PBX systems. G.729 is available for bandwidth-constrained deployments. WebRTC endpoints are supported with Opus codec, transcoded to G.711 at the trunk boundary. TLS + SRTP encryption is available for secure media transport.
Can I use Talkloo SIP trunking with an AI voice agent?
Yes - and this is a native feature, not an integration project. Talkloo's SIP trunk voice AI capability lets you bind a SIP trunk directly to a Talkloo AI voice agent. Inbound calls on a DID are answered by the AI agent automatically. Outbound AI calling campaigns use the same trunk. No middleware, no separate CPaaS account, no webhook plumbing. It's programmable SIP with AI built in.
What happens if my SIP trunk goes down - is there failover?
Yes. Talkloo's hosted SIP trunk infrastructure includes automatic failover routing. If a primary route degrades or a carrier segment fails, calls re-route to backup carriers without manual intervention. The platform targets a 99.9% uptime SLA. You can also configure Talkloo as a secondary trunk behind your existing carrier - it costs nothing unless calls actually route through it.