Operations

Operations

This page collects operational commands and checks used when running GABPBX.

Service Control

Systemd deployments commonly use:

systemctl stop gabpbx
systemctl start gabpbx
systemctl status gabpbx
journalctl -u gabpbx

For crash debugging, run GABPBX under gdb instead of the service manager.

CLI

Start foreground console:

gabpbx -vvvc

Connect to a running instance:

gabpbx -r

Run one command:

gabpbx -rx "core show version"

Useful commands:

core show version
core show channels
core show channel <name>
module show
module show like <name>
logger reload
dialplan reload
manager show connected
rtp set debug on
rtp set debug off
rtp set debug ice on
rtp set debug ice off

rtp set debug ice {on|off} toggles ICE diagnostics independently of the general RTP debug. When enabled it logs candidate-pair nomination and the media source selected after ICE completion, which is useful when troubleshooting WebRTC media (one-way audio, audio that arrives late, or media reaching the wrong port).

For chan_sofia:

sip show settings
sip show version
sip show peers
sip show peer <peer>
sip show peer <peer> detail
sip show channels
sip show inuse
sip set debug on
sip set debug peer <peer>
sip set debug ip <addr>
sip set debug sdp on
sip set debug sdp off
sip set debug fork on
sip set debug fork off
sip set debug capture <host:port>
sip set debug file <path>
sip show blacklist

Sofia-SIP library version

sip show version reports which Sofia-SIP library chan_sofia is using:

*CLI> sip show version

Sofia-SIP:
----------
  Built against:   2.0.4 (sofia-sip-2.0.4)
  Runtime version: sofia-sip-2.0.4
  Loaded library:  /usr/local/lib/libsofia-sip-ua.so.0

• Built against is the compile-time version from the Sofia-SIP headers the driver was built with.
• Runtime version is a symbol exported by the shared object actually loaded, so a value that differs from Built against means the running library is newer or older than those headers — rebuild chan_sofia to match.
• Loaded library is the resolved path of that shared object (the ABI-stable soname link).

SDP / ICE tracer

sip set debug sdp {on|off} is a focused SDP and ICE negotiation tracer, off by default and independent of the general sip set debug. When enabled it dumps:

• the generated offer SDP,
• the WebRTC offer-gate decision (whether DTLS-SRTP is offered for the target, and why),
• the incoming answer SDP, and
• the offerer answer-apply path.

Because it traces inside chan_sofia, it shows the signaling of a WSS or TLS session already decrypted — something a tcpdump/sngrep capture on the wire cannot give you. It is the first tool to reach for when an inbound WebRTC call (GABPBX as the SDP offerer) negotiates audio/video incorrectly.

Call-fork tracer

sip set debug fork {on|off} is a focused tracer for parallel call forking, off by default and independent of the general sip set debug. When a single address of record has several bound contacts (for example a physical phone over UDP and a browser over WSS on the same account), a call forks a leg to each contact. The tracer logs each leg's creation, the selection of the answering ("winner") leg, and the teardown of the losing legs. It is the clearest way to follow a mixed-transport fork and to spot a leg being torn down against a stale or removed contact.

Transaction tag on debug lines

Every Sofia: … debug line now carries a [from-user|to-user|callid] tag, so a log line can be attributed to a specific SIP transaction even when several calls are in flight at once. Filter the log on a Call-ID prefix or a user to follow one call through a busy trace.

Inspecting a peer

sip show peer <peer> prints a concise, readable summary of a single peer:

• SIP peer name
• Registration state
• Endpoint (user@host:port and the transport in use)
• Context / source (static config or realtime)
• Accountcode
• Contact slots used / allowed
• Media (codecs, DTMF mode, NAT, directmedia)
• Calls (active / limit, ringing, on-hold)
• Qualify status

The summary is followed by the full list of bound contacts, each showing its State, TTL, Source address, and User-Agent.

sip show peer <peer> detail prints the complete per-field dump for the peer, grouped into sections: session timers, identity headers, network & media, limits & features, fax / T.38, timers & RTP, routing & dialplan, security & ACL, registration, and contacts. The detail keyword tab-completes.

Logging

Main config:

configs/logger.conf.sample

Reload:

*CLI> logger reload

When debugging SIP, collect:

• GABPBX console logs.
• sip set debug output.
• Packet capture with tcpdump or sngrep.
• AMI events if monitoring behavior is involved.

Packet Capture

Examples:

tcpdump -i any -n -s 0 -w /tmp/gabpbx-sip.pcap udp port 5060
tcpdump -i any -n -s 0 -w /tmp/gabpbx-rtp.pcap portrange 10000-20000

Use sngrep for SIP ladder inspection when available.

Live SIP Capture (HEP)

chan_sofia can export a copy of every SIP message it sends or receives — already decrypted, including the SDP body, and covering every transport (udp, tcp, tls, ws, wss) — to a capture sink. This gives the wire view that a tcpdump capture cannot provide for TLS or WebSocket transports, because the messages are exported after decryption.

Two sinks are available and may be used together.

Stream to a Homer / sipcapture (HEP) collector, configured in the [general] section of sofia.conf:

[general]
sip_capture_address = 127.0.0.1:9060

Write the same decrypted stream to a plain-text file:

[general]
sip_capture_file = /var/log/gabpbx/sip_capture.txt

Either sink can also be toggled at runtime from the CLI, without a reload:

sip set debug capture <host:port>
sip set debug file <path>

Read the HEP stream with an EEP/HEP-capable sngrep listening on the same address:

sngrep -L udp:127.0.0.1:9060

Build Hygiene

Normal build:

./configure
make
make install

Avoid make clean or make dist-clean unless intentionally resetting build state, because it forces much more recompilation.

Git Hygiene

For the public GitHub repository:

• Keep main protected.
• Work through branches and pull requests for public changes.
• Do not commit local memory/private documentation.
• Keep generated build artifacts out of the repository.
• Keep source files needed to build from a fresh clone tracked.

default.exports is a required source/build support file and must be tracked.

SIP Production Checklist

• Disable guest calls unless required.
• Use strong peer secrets.
• Configure NAT: externaddr, localnet, nat=force_rport,comedia.
• Keep directmedia=no until RTP is proven.
• Configure call limits on trunks.
• Disable REFER transfers on untrusted trunks.
• Monitor authentication failures and blacklist entries.
• Verify RTP both directions before and after transfers.
• Verify codecs on both call legs.
• Restart GABPBX when switching SIP channel drivers.