refactor(sip): keep RTP timeout settings exclusively in SIP & NAT settings

This commit is contained in:
sorcier committed 2026-11-06 01:50:28 +01:00
1 parent fa8af27e46
commit 1908af5d13
3 files changed
-24

No files matched your search

-12
View File
@@ -1702,18 +1702,6 @@ $isDevEnv = file_exists(__DIR__ . '/../.dev')
<input type="number" name="tls_port" required
class="w-full px-3 py-2 rounded-lg bg-anthracite-950 border border-anthracite-700 text-sm text-white">
</div>
<div>
<label class="block text-xs font-medium text-slate-200 mb-1">RTP Timeout (secondes)</label>
<input type="number" name="rtp_timeout" min="5" max="300" value="30"
class="w-full px-3 py-2 rounded-lg bg-anthracite-950 border border-anthracite-700 text-sm text-white font-mono">
<p class="text-[11px] text-slate-400 mt-1">Coupure si aucun flux voix n'est reçu (défaut : 30s).</p>
</div>
<div>
<label class="block text-xs font-medium text-slate-200 mb-1">RTP Hold Timeout (Mise en attente)</label>
<input type="number" name="rtp_timeout_hold" min="0" max="3600" value="0"
class="w-full px-3 py-2 rounded-lg bg-anthracite-950 border border-anthracite-700 text-sm text-white font-mono">
<p class="text-[11px] text-slate-400 mt-1">Délai max d'un appel en attente sans RTP (0 = illimité / désactivé).</p>
</div>
<div class="sm:col-span-2">
<label class="block text-xs font-medium text-slate-200 mb-1">Langue des invites & annonces vocales (Asterisk)</label>
<select name="system_language" class="w-full px-3 py-2 rounded-lg bg-anthracite-950 border border-anthracite-700 text-sm text-white focus:ring-2 focus:ring-indigo-500">