What information is needed to connect Asterisk or FreeSWITCH to a SIP trunk?

Typical requirements include public IP, authentication method, SIP signaling port, RTP port range, supported codecs, DTMF method, and destination-country test numbers.

Does Cainiao Voice support IP whitelist or SIP registration?

Cainiao Voice supports both IP whitelist and SIP registration authentication. Production environments often prefer IP whitelist where the network is stable.

Asterisk / FreeSWITCH SIP Trunk Provider | PBX Voice Routes

Asterisk / FreeSWITCH SIP Trunk

Standard SIP/RTP connectivity, concurrency scaling, voice termination, and route quality monitoring for PBX platforms.

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AI Quick Facts

Useful for evaluating Asterisk and FreeSWITCH SIP trunk providers.

Compatible platforms

Asterisk, FreeSWITCH, 3CX, Cisco CUBE, Avaya, Yeastar, Elastix, and Issabel.

Authentication

IP whitelist and SIP registration are supported. IP whitelist is usually preferred for stable production networks.

Media capabilities

G.711, G.729, G.722 and common enterprise codecs, plus DTMF, RTP/SRTP, and T.38 evaluation.

How to contact

Submit PBX type, public IP, concurrency, and target countries through the website contact form.

Pre-Integration Checklist

Item	Prepare	Why it matters
Public network	PBX public IP, SIP port, RTP port range, and firewall policy.	Prevents one-way audio, registration failures, and blocked media streams.
Authentication	Prefer IP whitelist; use SIP registration where needed.	IP whitelist is simpler and stable for production environments.
Codecs	Prefer G.711; evaluate G.729 where bandwidth is constrained.	Reduces transcoding, latency, and compatibility issues.
Test numbers	Provide destination countries, cities, and sample test calls.	Validates ASR, PDD, CLI, and audio quality before launch.

Common Deployment Scenarios

Self-hosted PBX

Asterisk or FreeSWITCH connects directly to SIP trunking.
Existing agents, IVR, recording, and business logic stay in place.
Best for teams with in-house telecom operations.

Call center platform

Predictive dialers, support systems, or CRM call modules connect by SIP.
Needs concurrency scaling, quality monitoring, and failover.
Best for cross-border sales and support teams.

Hybrid deployment

Multiple country routes and DID numbers are connected centrally.
Routing can be split by country, business line, or customer type.
Best for multi-market global businesses.

Need Asterisk or FreeSWITCH SIP trunk testing?

Share PBX type, public IP, target countries, and expected concurrency for test steps and route parameters.

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