I will twilio sip trunk 3cx configure inbound calling outbound calling did nuumber
Professional VoIP PBX Call Center Expert
About this Gig
Hi! I will help you set up and troubleshoot VoIP calling systems using SIP Trunks & popular PBX platforms like 3CX, FreePBX, & Asteriskincluding inbound call routing, caller ID, IVR, and integration with Twilio (if needed).
If youre dealing with no audio, call not connecting, registration issues, wrong caller ID, inbound not routing, or IVR problems, youre in the right place.
What I can do for you
- SIP trunk setup (registration, authentication, codecs)
- Inbound call routing to extensions/queues
- Caller ID config (format, validation, reliable sending)
- IVR setup (menus, transfers, DIDs)
- 3CX setup / FreePBX configuration / Asterisk configuration
- Twilio integration for SIP/Calling
Troubleshooting & fixes for:
- Failed registration / credentials issues
- One-way audio (audio only in one direction)
- No answer / calls not ringing
- Codec mismatch / RTP issues
- Inbound calls not reaching destination
What I need from you
PBX type/version (3CX/FreePBX/Asterisk), SIP trunk provider, DID numbers (if inbound), & your current settings/screenshots (or logs).
After setup, I test calling scenarios & deliver a clear summary of what was changed.
Reply now with your provider + PBX type to get started
FAQ
Do you need admin access?
Usually you can share configuration screenshots/logs. If you can grant controlled access, we’ll do it safely.
Which providers do you support? (Telnyx / Twilio / generic SIP?)
I can integrate with most standard SIP trunk providers. If you share your provider requirements (auth/codec/DID format), I’ll adapt.
Can you fix one-way audio or RTP issues?
Yes—those are common and usually solvable with codec, NAT, RTP port and firewall checks.
Will inbound calls ring my extensions/queues?
That’s the goal. I’ll configure inbound rules and test routing end-to-end.
