I will configure sip trunk 3cx freepbx asterisk voip troubleshoot call routing issues
VoIP SIP Trunk and Telephony Specialist Fixing Broken Phone Systems Fast
About this Gig
SIP trunk or VoIP calls failing? I configure, troubleshoot and optimize business phone systems built around SIP trunking, 3CX, FreePBX, Asterisk and IP-PBX environments.
I can help with SIP registration, inbound/outbound calls, DID routing, caller ID, IVR, extensions, codecs, NAT/firewall issues, one-way audio, failed calls, routing errors and trunk connectivity.
Services include:
- SIP trunk setup and provider configuration
- 3CX, FreePBX and Asterisk configuration
- Inbound/outbound call routing and DID mapping
- SIP registration and authentication troubleshooting
- One-way/no audio, NAT and firewall troubleshooting
- IVR, extensions, ring groups and basic call flows
- Call testing and post-configuration verification
- VoIP migration and troubleshooting
I troubleshoot from the actual symptoms and logs instead of random settings. If your trunk registers but calls fail, calls connect with no audio, inbound numbers do not route, or your PBX cannot reach the provider, I can work through the fault systematically.
Platforms/providers may include 3CX, FreePBX, Asterisk, Twilio and Telnyx.
For complex issues, send your current setup or error details before ordering.
Provider:
VICIdial
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3CX
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Asterisk
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FreePBX
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Twilio
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Mircosoft Teams
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Other
My Portfolio
FAQ
Can you set up a SIP trunk on 3CX, FreePBX or Asterisk?
Yes. I can configure the SIP trunk, authentication, transport settings, DID mapping, inbound/outbound routes and call testing. The exact configuration depends on your SIP provider, PBX version, network environment and provider requirements.
My SIP trunk is registered but outbound calls fail. Can you fix it?
Yes. I can investigate outbound routing, dial patterns, authentication, provider restrictions, caller ID, codec/transport settings, firewall/NAT behavior and related SIP responses to isolate where the call is failing.
Can you fix one-way audio or calls with no audio?
Yes. One-way or missing audio can involve RTP ports, NAT, firewall rules, codec negotiation or incorrect network configuration. I can inspect the relevant SIP and media configuration and test the call path.
Can you configure inbound calls and DID numbers?
Yes. I can configure DID routing so incoming numbers reach the correct extension, ring group, IVR, queue or other destination supported by your PBX and provider.
Can you troubleshoot SIP registration errors?
Yes. I can troubleshoot authentication failures, registration rejection, transport settings, DNS issues, provider-side requirements, incorrect credentials and connectivity problems.
Do you work with Twilio or Telnyx SIP trunks?
Yes, when the required services and account configuration are available. I can work with SIP trunking and call-routing setups involving providers such as Twilio and Telnyx, alongside supported PBX platforms.
Can you troubleshoot an existing VoIP system instead of building a new one?
Absolutely. In many cases troubleshooting an existing system is the better approach. I can work from your current configuration, logs, symptoms and call behavior instead of rebuilding everything unnecessarily.
Can you configure IVR, extensions and call routing?
Yes. Depending on the package and complexity, I can configure extensions, inbound routes, outbound routes, ring groups, IVR menus and basic call flows within the supported PBX environment.
Can you fix VoIP problems caused by NAT or firewall settings?
Yes. NAT and firewall issues are common causes of SIP and RTP problems. I can review the relevant network and PBX settings and test signaling and media flow to identify configuration conflicts.
Can you migrate or improve an existing SIP/VoIP setup?
Yes. I can help with selected VoIP migrations, SIP trunk replacements, PBX changes and configuration improvements. For larger migrations involving many numbers, complex routing or production cutovers, message me first so the scope can be assess

