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diagnostics: System → Diagnostics with a network test probe; RTP_PORT_START/END

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The server checks itself (public IP, DNS, local networks, phone ACL, SIP transports, free audio
ports) and says how to fix each problem. A quick or complete network test asks the probe
(apps/probe, own image) to test this server from outside: HTTP/HTTPS, SIP over UDP/TCP/TLS, the
browser phone's /ws, sampled audio ports. The probe only tests the address that asks.
The audio port range comes from RTP_PORT_START/END (default 10000-20000).
This commit is contained in:
iamdoubz committed 2026-10-21 12:37:16 -05:00
1 parent 90b3081368
commit 65a0438dde
50 files changed
+1704 -59
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apps/api/test/asterisk/call-test.ts
+31
View File
@@ -23,6 +23,7 @@ const NET = 'sp-test';
const AST = 'sp-test-ast';
const PHONES = 'sp-test-phones';
const PN_PHONE = 'sp-test-pn-phone';
const RTP_PORTS = { start: 31000, end: 31200 };
const AMI_PORT = 15038;
const AMI_SECRET = 'call-test-secret';
const PASS = { '101': 'CallTest101xxxxxxxxxxxxx', '102': 'CallTest102xxxxxxxxxxxxx' };
@@ -79,6 +80,8 @@ function snapshot(): Snapshot {
announcements: [],
recordingAnnouncement: null,
extensions: [ext('101', 'Alice'), ext('102', 'Bob')],
// Not rtp.conf's 10000–20000: the generated range must win (RTP_PORT_START/END).
rtpPorts: RTP_PORTS,
};
}
@@ -174,6 +177,13 @@ async function main() {
ami.start();
await new Promise((r) => ami.once('connected', r));
log('AMI connected from the host');
const rtp = await ami.command('rtp show settings');
if (
!new RegExp(`Port start:\\s+${RTP_PORTS.start}\\b`).test(rtp) ||
!new RegExp(`Port end:\\s+${RTP_PORTS.end}\\b`).test(rtp)
)
problems.push(`RTP ports aren't ${RTP_PORTS.start}–${RTP_PORTS.end}:\n${rtp}`);
else log(`RTP ports ${RTP_PORTS.start}–${RTP_PORTS.end} (generated, over rtp.conf's)`);
if (RECORD) {
// A second, raw AMI session so the fixture holds exactly what Asterisk sent.
recorder = new Socket();
@@ -237,6 +247,27 @@ async function main() {
).out.trim();
if (!wsReg.startsWith('SIP/2.0 200')) problems.push(`102-web over WebSocket: ${wsReg}`);
else log('102-web over WebSocket from 127.0.0.1: registered');
// The network test probe's SIP checks (08 §2.6): Asterisk answers its OPTIONS, UDP and TCP.
const probed = docker(
'run',
'--rm',
'--network',
NET,
'-v',
`${join(import.meta.dirname, '../../../probe/src')}:/probe:ro`,
'node:24-trixie-slim',
'node',
'--input-type=module',
'-e',
`import { runTest } from '/probe/checks.ts';
for (const kind of ['sip-udp', 'sip-tcp'])
console.log(JSON.stringify(await runTest('${AST}', 'pbx.test', { id: kind, kind, port: 5060 })));`,
).out;
for (const kind of ['sip-udp', 'sip-tcp'])
if (!new RegExp(`"id":"${kind}","ok":true,"detail":"SIP/2.0 \\d{3}`).test(probed))
problems.push(`probe ${kind}: ${probed.trim()}`);
if (!problems.some((p) => p.startsWith('probe ')))
log(`probe: Asterisk answers SIP OPTIONS over UDP and TCP`);
// 101 also on an iOS Linphone: '&' in its push params used to break Dial for every device.
const pn = docker(
'run',