A free, browser-based toolkit for VoIP engineers. Analyze logs, inspect packet captures, and debug SIP calls β no software installation required.
20
Tools available
100%
Browser-based
0
Data uploaded to cloud
Available tools β Ώ drag to reorder
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Dial Pattern Creator
β ReadyNew
Interactively build FreePBX outbound route dial patterns. Configure local, long-distance, international, toll-free, mobile, and emergency rules for your country β with trunk prefix stripping and prepend logic.
Parse and visualize FreeSWITCH log files. Groups by call UUID, shows call flows with channel states, module-level log grouping, bridge detection, and hangup cause analysis.
Parse and visualize FreePBX system logs (freepbx.log). See the system event timeline, module errors, reload events, auth failures, and filter by level or module.
Upload a .pcap or .pcapng file and get an instant VoIP health report: auth failures, retransmissions, NAT issues, codec mismatches, and RTP quality metrics.
Visualize SIP call flows from PCAP files as sequence diagrams β just like Wireshark's Flow Graph, but in your browser. Click any message to inspect the raw SIP.
Paste any raw SIP message (INVITE, 200 OK, BYE, REGISTERβ¦) and get a color-coded breakdown of every header. Validates structure and highlights potential issues.
Paste sngrep exports or raw SIP messages and get an instant SIP ladder diagram with call summary. No PCAP needed β works directly from copied terminal output.
Test SIP extension registration live against any server. Detects auth failures (401/403), measures registration RTT, identifies realm misconfiguration, and shows the full SIP exchange step by step.
Paste a 401 Unauthorized challenge + credentials to verify if the digest response is correct. Calculates HA1, HA2, and final MD5 step by step. Compares computed vs sniffed Authorization headers.
Paste an SDP and simulate how RTP flows behind NAT. Visualizes the media path, detects private IPs leaking into SDP, identifies one-way audio risk, and suggests Asterisk/FreeSWITCH fixes.
Give two SDPs (offer + answer) and see exactly which codec was chosen, why others were rejected, the priority order from both sides, and DTMF negotiation result. Detects mismatches that cause 488 errors.
Search any Q.850, SIP, Asterisk, or FreeSWITCH hangup cause and get a human explanation, common root causes, and actionable fixes. E.g. "Cause 34 β No circuit available β trunk saturated".
Paste a call alert log and instantly detect trampoline loops, retry storms, and duplicate CDR entries. Visualizes call attempts by gateway over time and flags alternating-gateway patterns.