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English
$
USD
I will troubleshoot fusionpbx and freeswitch sip call issues
About this Gig
Are you experiencing SIP registration failures, one-way audio, dropped calls, no-ring issues, incorrect call routing, or trunk connectivity problems in FusionPBX or FreeSWITCH?
I will diagnose and troubleshoot your VoIP/PBX environment and provide a practical, reliable solution. I work with FusionPBX, FreeSWITCH, Asterisk, FreePBX, Twilio, SIP trunks, extensions, inbound and outbound routing, NAT/firewall settings, caller ID, codecs, and call-quality issues.
Depending on the selected package, I can:
- Review SIP profiles, gateways, trunks, extensions, dialplans, and call routes
- Analyze logs and SIP signaling
- Fix registration, authentication, NAT, firewall, RTP, and one-way-audio problems
- Test inbound and outbound calls
- Explain the root cause and document the changes
Please contact me before ordering if the scope involves multiple servers, custom development, high-availability architecture, carrier migrations, or predictive dialers.
Provider:
Asterisk
•
FreePBX
•
Twilio
•
Other
FAQ
What access will you need?
Usually SSH access and FusionPBX admin access. I will confirm the minimum required access before work begins.
Can you work with an existing production server?
Yes. I first review the issue and current setup, and I avoid unnecessary changes. Please ensure you have a recent backup.
