Merge pull request #232 from outis1one/claude/group-call-busy-signal-e8usba
Fix inbound PSTN calls: dialplan context never loaded, Caller-ID never real
This commit is contained in:
Outis
@@ -2,9 +2,14 @@
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This is the exact sequence that got a real Anveo Direct DID working end to
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end (both outbound and inbound) with `asterisk-digital-ocean.sh` +
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`pstn-trunk.sh`, confirmed live on a real droplet. Follow it in order for
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each additional number — steps 1–2 are one-time account setup; steps
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3–7 repeat per DID.
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`pstn-trunk.sh`, confirmed live on a real droplet.
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**Steps 1, 3 and 4 are one-time account setup** — the outbound Service
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Trunk (step 3) and the inbound SIP Trunk (step 4) each cover every DID on
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the account, no matter how many you add. **Only steps 2, 4.5, 6 and 7
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repeat per number**: order the DID, point it at the existing inbound trunk
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(skippable entirely if you set the account default in step 4.6), assign it
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in the dashboard, and test.
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## 0. Prerequisites
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@@ -91,7 +96,7 @@ catch-all entry (prefix `1`); a few specific area codes have their own
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slightly different override rates, and Virgin Islands is billed
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separately/higher (~$0.02/min) despite looking like a normal US number.
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## 4. Create the SIP Trunk (inbound forwarding) — once per DID
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## 4. Create the SIP Trunk (inbound forwarding) — ONE covers every DID
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Anveo has **two unrelated "trunk" concepts** — don't confuse them:
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- The **Outbound Service Trunk** from step 3 handles calls *out*.
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@@ -99,11 +104,20 @@ Anveo has **two unrelated "trunk" concepts** — don't confuse them:
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routing objects live) handles calls *in* — this is what actually
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populates a DID's "Destination SIP Trunk" dropdown.
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For **each DID**, create one of these:
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**You only need ONE of these for the whole account**, no matter how many
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DIDs you have. The `$[E164]$` placeholder in step 2 is what makes it
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reusable — Anveo substitutes each DID's own number at call time, so the
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same trunk object forwards `15551111111@IP:5060` for one DID and
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`15552222222@IP:5060` for another. The Asterisk side matches whatever
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arrives (`_X.` in `[from-pstn-trunk]`) and looks it up in
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`pstn-personal-dids.conf` to decide who rings. Creating one per DID is
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possible but pointless unless different DIDs need to forward to genuinely
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different destinations.
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1. **Trunk Name**: any label (e.g. `asterisk-do-inbound`) — can reuse the
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same one for multiple DIDs if you want them all forwarding the same
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way, or make one per DID for clarity.
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Create it once (steps 1–4), then per DID only step 5 applies — or set the
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account default in step 6 and skip even that:
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1. **Trunk Name**: any label (e.g. `asterisk-do-inbound`).
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2. **Primary**: type **SIP URI**, value:
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```
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$[E164]$@<this box's public IP>:5060
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@@ -115,12 +129,13 @@ For **each DID**, create one of these:
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in the actual INVITE Anveo sends.
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3. **Failover**: leave blank.
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4. Save.
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5. Go to the DID's own **Call Options** tab → set **Destination SIP
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Trunk** to this new SIP Trunk object (not the Outbound Service Trunk)
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5. **Per DID**: go to that DID's **Call Options** tab → set **Destination
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SIP Trunk** to this SIP Trunk object (not the Outbound Service Trunk)
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→ Save.
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6. **Optional, recommended once you're adding more DIDs**: Account
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6. **Recommended if you have more than one or two DIDs**: Account
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Options → Service Defaults → set **Default Destination Trunk** to this
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SIP Trunk, so future DIDs auto-route here without repeating step 5.
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SIP Trunk. New DIDs then auto-route here and step 5 becomes unnecessary
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entirely.
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## 5. Configure the droplet
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@@ -216,13 +231,56 @@ installer, not configuration mistakes — listed here for context on what
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under `setup.sh`'s `set -u` on `source`. Now guarded and falls back to
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safe defaults, erroring clearly only if genuinely required fields are
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missing.
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- **The inbound `[from-pstn-trunk]` context never loaded at all** — it used
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to share one `#include`'d file with the outbound dialplan, where the file
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continued `[intercom]` first and only declared `[from-pstn-trunk]`
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partway through. Asterisk silently failed to load the *entire* file
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(outbound patterns included) with no error anywhere, so every inbound
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call got "extension not found in context 'from-pstn-trunk'" and an
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immediate SIP rejection. Inbound now lives in its own file that starts
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with its context header, matching the structure of the messaging
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dialplan that always loaded correctly.
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- **Every inbound call's Caller-ID arrived as the DID itself**, not the
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real caller — the trunk endpoint pinned a static `callerid=<DID>` and
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didn't set `trust_id_inbound`, so Asterisk ignored the identity Anveo
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actually sends and fell back to that default. Every restricted-tier
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`allowed_numbers` check was therefore unwinnable regardless of config.
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Fixed by dropping the static default (redundant — the outbound dialplan
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sets `CALLERID(num)` on the channel itself) and adding
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`trust_id_inbound=yes`.
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- **A leading `+` broke whitelist matching** — once the fix above started
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surfacing real caller identity, Anveo delivers it `+E.164` style
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(`+15551234567`). The normalizer only added a leading `1` to 10-digit
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values and never stripped the `+`, so a 12-character value could never
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match an 11-digit, digits-only `allowed_numbers` entry no matter how
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correctly the number was whitelisted. The `+` is now stripped in the
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dialplan, the group-ring script, and the dashboard's admin-input side
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(so pasting a number straight from a call log works too).
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- **Bare `Hangup()` on inbound denials** instead of `Busy(15)` — three
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inbound denial paths didn't produce a proper "486 Busy Here", making a
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correct permission denial sound like a generic call failure and
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disguising config mismatches as dialplan bugs.
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## Dashboard changes may need "Commit Changes"
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Confirmed live: edits made in the Security Dashboard's PSTN Trunk tab
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(permission tiers, groups, personal-DID owners, limits) are written to disk
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immediately, but `AST_CONFIG()` in the dialplan has been observed still
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returning the *old* value until the Asterisk container is fully restarted —
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a `dialplan reload` / `module reload` is not enough. This contradicts the
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"reads fresh on every call, no restart needed" design premise these files
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otherwise rely on; root cause not established.
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The PSTN Trunk tab has a **Commit Changes (Restart Asterisk)** button for
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this, which appears after any save on that tab and warns if you navigate
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away with changes uncommitted. Existing calls are unaffected by the
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restart. If a change doesn't seem to be taking effect on a test call, press
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it before assuming the config itself is wrong.
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## Still open
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- The 2-phone-number account cap (see step 1) blocks getting all 4
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planned numbers until either Anveo lifts it or a tax ID is provided —
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tabled for now.
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- The interactive CLI walkthrough (`pstn-trunk.sh` actually prompting
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through account setup step by step, not just this static doc) hasn't
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been built yet — this guide is the reference for building that once
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there's appetite for it.
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- Nothing blocking. The two items previously listed here are both resolved:
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the 2-phone-number account cap no longer applies (additional DIDs have
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since been obtained), and the interactive CLI walkthrough was built —
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see `_pstn_anveo_walkthrough` in `services/pstn-trunk.sh`, reachable from
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the provider quick-pick when you choose Anveo Direct.
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+27
-23
@@ -583,30 +583,30 @@ __ALERT_KILLED_IN_LINE__
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; live config files and applies the identical tier/allowed_numbers logic
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; in a plain shell loop.
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;
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; Every OTHER denial path in this file (outbound tier/number/intl/killswitch,
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; the shared ring-group's pstn_in_denied, this owner's own denial just
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; above) plays a proper Busy(15) tone before Hangup() — pstn_in_denied and
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; the personal-DID denial paths were missing that Busy() and fell straight
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; to a bare Hangup() on an unanswered channel instead, which chan_pjsip
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; does NOT map to "486 Busy Here" the way Busy() does. NOT yet confirmed
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; live against a real call, but this is a real, reproducible inconsistency
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; (grep this file for "Hangup()" without a preceding Busy() and these three
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; exits were the only inbound ones missing it) and is a plausible cause of
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; a caller hearing one ring and then an ambiguous fast-busy/call-failed
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; tone on what looks like a correctly-authorized group member: if the
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; group/tier/allowed_numbers config has ANY mismatch against the test
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; caller ID, the call was always going to be denied here regardless of
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; this fix — the missing Busy() just made a config problem sound like a
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; dialplan problem. Fixed below (Busy(15) added to all three inbound
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; denial exits). If a group-assigned personal number still goes straight
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; to busy after this, the NoOp() logged right after the SHELL() call in
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; pstn_personal_group_ring prints the resolved group name, caller ID, and
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; PSTN_RING_LIST — check `asterisk -rx "core show channels verbose"` / the
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; full log for that line first; an empty list there means the config (not
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; the dialplan) is the next thing to check — confirm the calling number is
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; in allowed_numbers for at least one CURRENT group member whose tier is
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; Every denial path in this file (outbound tier/number/intl/killswitch, the
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; shared ring-group's pstn_in_denied, the personal-DID owner and group
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; denials) plays a proper Busy(15) tone before Hangup(). pstn_in_denied and
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; the two personal-DID denial paths used to fall straight to a bare
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; Hangup() on an unanswered channel instead, which chan_pjsip does NOT map
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; to "486 Busy Here" the way Busy() does — making a config mismatch sound
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; like a dialplan failure. Confirmed live (2026-07-24): a group-owned
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; personal DID is now fully working end to end, ringing every authorized
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; member simultaneously, after this plus three other fixes found in the
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; same session (see _pstn_write_inbound_dialplan_include's file-split
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; comment, and _pstn_write_pjsip_include's trust_id_inbound/callerid
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; comment).
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;
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; If a group-assigned personal number goes straight to busy, the NoOp()
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; logged right after the SHELL() call in pstn_personal_group_ring prints
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; the resolved group name, caller ID, and PSTN_RING_LIST — check the
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; Asterisk console/full log for that line first. An empty list there means
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; the config, not the dialplan: confirm the calling number is in
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; allowed_numbers for at least one CURRENT group member whose tier is
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; "restricted" or "full" in pstn-permissions.conf, and that the group's
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; members= line in pstn-groups.conf actually lists that extension.
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; members= line in pstn-groups.conf actually lists that extension. Note
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; that a dashboard edit may need the "Commit Changes" button (a container
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; restart — see restart_asterisk_container() in services/security-dashboard.sh)
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; before AST_CONFIG() actually returns the new value.
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exten => pstn_personal_inbound,1,GotoIf($["${PSTN_PERSONAL_OWNER:0:1}" = "@"]?pstn_personal_group_ring,1)
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same => n,Set(PSTN_OWNER_TIER=${AST_CONFIG(pstn-permissions.conf,${PSTN_PERSONAL_OWNER},tier)})
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same => n,GotoIf($["${PSTN_OWNER_TIER}" = "full"]?pstn_personal_ring,1)
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@@ -672,6 +672,10 @@ EOF
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# Safe REGEX direction: $allowed is admin-entered (pstn-permissions.conf),
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# $caller is the incoming Caller-ID — pattern is always the admin data,
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# string is always the caller-controlled data, never the reverse.
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# Confirmed live (2026-07-24): SHELL() invoking this script from the
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# dialplan works end to end against a real inbound call — the returned
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# &-joined list rings every authorized member simultaneously, first to
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# answer wins, and unauthorized members are correctly skipped.
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_pstn_write_personal_group_ring_script() {
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local FILE="$1"
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cat > "$FILE" << 'SCRIPT'
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@@ -458,15 +458,20 @@ _secdash_write_sudoers() {
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# `docker exec -i <container> tee <exact path>` instead of a direct
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# host-side file write (see _secdash_grant_asterisk_access's comment on
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# why), plus the two Asterisk CLI calls needed after a change and the
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# live registration-status check. All six are exact commands, no
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# live registration-status check. All seven are exact commands, no
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# wildcards, scoped to the one container actually installed on this box.
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# The last line (docker restart) backs the PSTN Trunk tab's "Commit
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# Changes" button — see restart_asterisk_container()'s comment for why
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# that exists (AST_CONFIG() live-reads not always picking up dashboard
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# edits without a full container restart).
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if [ -n "$_ea_container" ]; then
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_ea_lines="$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec -i $_ea_container tee /etc/asterisk/pjsip.conf
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec -i $_ea_container tee /etc/easy-asterisk/categories.conf
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec -i $_ea_container tee /etc/easy-asterisk/rooms.conf
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec $_ea_container asterisk -rx module\ reload\ res_pjsip.so
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec $_ea_container asterisk -rx pjsip\ show\ endpoints
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec $_ea_container /usr/local/bin/easy-asterisk --rebuild-dialplan"
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker exec $_ea_container /usr/local/bin/easy-asterisk --rebuild-dialplan
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/docker restart $_ea_container"
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fi
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cat > /etc/sudoers.d/security-dashboard << SUDOERS
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$_svc_user ALL=(root) NOPASSWD: /usr/bin/cscli decisions delete --id [0-9]*
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@@ -1644,6 +1649,27 @@ def ea_rebuild_dialplan():
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run_sudo(["docker", "exec", ASTERISK_EA_CONTAINER, "/usr/local/bin/easy-asterisk", "--rebuild-dialplan"])
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def restart_asterisk_container():
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"""Restarts the Easy Asterisk container - the "Commit Changes" button on
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the PSTN Trunk tab. Confirmed live: dashboard writes to
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pstn-permissions.conf/pstn-groups.conf/pstn-personal-dids.conf land on
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disk immediately (readable via a plain `cat` right after saving), but
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AST_CONFIG() in the dialplan sometimes kept returning a stale value
|
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until the container was fully restarted - not just a `dialplan reload`
|
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or `module reload`, an actual container restart. Root cause not fully
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understood (contradicts AST_CONFIG's whole "reads fresh every call, no
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restart needed" design premise, which this codebase otherwise relies on
|
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throughout), but the restart reliably clears it, so this button exists
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instead of requiring every admin to rediscover "just restart it" the
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hard way. Uses the same ASTERISK_EA_CONTAINER/run_sudo mechanism as the
|
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Easy Asterisk Admin tab's own docker exec calls - no new sudoers scope
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needed beyond the one line added for this."""
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if not ASTERISK_EA_CONTAINER:
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return False, "No Asterisk container detected on this box"
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ok, _out, err = run_sudo(["docker", "restart", ASTERISK_EA_CONTAINER], timeout=30)
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return ok, ("" if ok else (err or "Restart failed"))
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|
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def ea_get_status():
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"""Registered/unregistered per extension — same 'pjsip show endpoints'
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parsing as the vendored get_registered_endpoints()."""
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@@ -2311,9 +2337,17 @@ INDEX_HTML = """<!doctype html>
|
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</div>
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</div>
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<div id="tab-pstn" style="display:none">
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<div class="card" id="pstn-restart-banner" style="display:none; border-left:4px solid #d9822b">
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<b>Unsaved changes may not be live yet.</b>
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<p class="muted" style="margin:0.25rem 0 0.5rem">
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Edits here are written to disk immediately, but Asterisk doesn't always pick them up without a restart — confirmed on personal-DID group reassignment specifically. Existing calls are never affected.
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</p>
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<button class="action" id="pstn-restart-btn">Commit Changes (Restart Asterisk)</button>
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<span id="pstn-restart-msg" class="muted" style="margin-left:0.5rem"></span>
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</div>
|
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<div class="card">
|
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<h3 style="margin-top:0">Concurrent-call caps</h3>
|
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<p class="muted">A call over either cap gets a busy signal (and an ntfy alert, if enabled) — existing calls are never affected. Changes apply live, on the next call.</p>
|
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<p class="muted">A call over either cap gets a busy signal (and an ntfy alert, if enabled) — existing calls are never affected. Changes are usually live on the next call; if a call doesn't reflect a recent change, use "Commit Changes" below.</p>
|
||||
<div class="row">
|
||||
<label class="muted" style="white-space:nowrap">Max outbound<br><input type="text" id="limit-out" style="width:5rem"></label>
|
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<label class="muted" style="white-space:nowrap">Max inbound<br><input type="text" id="limit-in" style="width:5rem"></label>
|
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@@ -2327,7 +2361,7 @@ INDEX_HTML = """<!doctype html>
|
||||
<b>internal</b> — no PSTN, can still call/receive other extensions and internal ring groups.
|
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<b>restricted</b> — internal, plus only pre-approved US numbers.
|
||||
<b>full</b> — internal, plus any US number.
|
||||
Changes apply live, on the next call — no Asterisk restart needed.
|
||||
Changes are usually live on the next call; if one doesn't seem to be taking effect, use "Commit Changes" below.
|
||||
</p>
|
||||
<p class="muted">
|
||||
<b>Messaging</b> — the same internal SIP texting flag as the Extensions tab's checkboxes, independent of the calling tier; this column is just a convenience for setting it alongside tier/numbers on one row. Enforced live by a dedicated dialplan context — see services/asterisk-digital-ocean.sh's README for how, and its caveat on the sender-extraction logic still needing real-traffic confirmation.
|
||||
@@ -2345,7 +2379,7 @@ INDEX_HTML = """<!doctype html>
|
||||
<div class="card">
|
||||
<h3 style="margin-top:0">Personal numbers</h3>
|
||||
<p class="muted">
|
||||
Multiple DIDs can share this one trunk. Assigning a DID to an extension routes inbound calls to that DID straight to its owner (still gated by the owner's own tier/approved-numbers above — no ring-group fallback), and makes that extension's outbound calls show this DID as Caller-ID instead of the shared trunk DID. You can also assign a DID to a <b>group</b> instead of a single extension — every current member whose own tier/approved-numbers authorize the caller rings (checked fresh on every call, so membership changes apply immediately); a group has no single extension to hang the outbound Caller-ID override on, so that part only applies to single-extension assignments. The shared DID/ring-group keeps working regardless.
|
||||
Multiple DIDs can share this one trunk. Assigning a DID to an extension routes inbound calls to that DID straight to its owner (still gated by the owner's own tier/approved-numbers above — no ring-group fallback), and makes that extension's outbound calls show this DID as Caller-ID instead of the shared trunk DID. You can also assign a DID to a <b>group</b> instead of a single extension — every current member whose own tier/approved-numbers authorize the caller rings, checked fresh against the group's current membership on every call; a group has no single extension to hang the outbound Caller-ID override on, so that part only applies to single-extension assignments. The shared DID/ring-group keeps working regardless. Reassigning a DID's owner has been confirmed to sometimes need "Commit Changes" (below) before Asterisk actually uses the new owner.
|
||||
</p>
|
||||
<div class="row">
|
||||
<input type="text" id="pd-did" placeholder="DID, e.g. 5551234567 (10 digits, no leading 1)" style="width:12rem">
|
||||
@@ -3118,6 +3152,7 @@ document.getElementById("grp-save").addEventListener("click", async () => {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("grp-msg").textContent = data.message || (data.ok ? "Saved" : "Failed");
|
||||
if (data.ok) markPstnDirty();
|
||||
loadGroups();
|
||||
});
|
||||
|
||||
@@ -3129,6 +3164,7 @@ async function applyGroupMessaging(name, enabled) {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("grp-msg").textContent = data.message || (data.ok ? "Applied" : "Failed");
|
||||
if (data.ok) markPstnDirty();
|
||||
loadMessaging();
|
||||
}
|
||||
|
||||
@@ -3140,6 +3176,7 @@ async function deleteGroup(name) {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("grp-msg").textContent = data.message || (data.ok ? "Deleted" : "Failed");
|
||||
if (data.ok) markPstnDirty();
|
||||
loadGroups();
|
||||
}
|
||||
|
||||
@@ -3152,9 +3189,41 @@ document.getElementById("limits-save").addEventListener("click", async () => {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("limits-msg").textContent = data.message || (data.ok ? "Saved" : "Failed");
|
||||
if (data.ok) markPstnDirty();
|
||||
loadPstnLimits();
|
||||
});
|
||||
|
||||
// "Commit Changes" — see restart_asterisk_container()'s comment in app.py
|
||||
// for why this button exists: AST_CONFIG() live-reads of these PSTN config
|
||||
// files have been confirmed to sometimes stay stale (returning what was on
|
||||
// disk at last container start, not the freshly-saved value) until a full
|
||||
// container restart, contradicting the "no restart needed" premise the
|
||||
// rest of this tab's copy otherwise relies on. pstnDirty tracks whether
|
||||
// ANY save on this tab succeeded since the last restart (or page load),
|
||||
// shows a persistent banner, and warns on tab-close/navigation so a change
|
||||
// doesn't silently sit uncommitted.
|
||||
let pstnDirty = false;
|
||||
function markPstnDirty() {
|
||||
pstnDirty = true;
|
||||
document.getElementById("pstn-restart-banner").style.display = "";
|
||||
}
|
||||
window.addEventListener("beforeunload", (e) => {
|
||||
if (!pstnDirty) return;
|
||||
e.preventDefault();
|
||||
e.returnValue = "";
|
||||
});
|
||||
document.getElementById("pstn-restart-btn").addEventListener("click", async () => {
|
||||
const msg = document.getElementById("pstn-restart-msg");
|
||||
msg.textContent = "Restarting Asterisk…";
|
||||
const res = await fetch("/api/asterisk-restart", { method: "POST" });
|
||||
const data = await res.json();
|
||||
msg.textContent = data.message || (data.ok ? "Restarted" : "Failed");
|
||||
if (data.ok) {
|
||||
pstnDirty = false;
|
||||
document.getElementById("pstn-restart-banner").style.display = "none";
|
||||
}
|
||||
});
|
||||
|
||||
let lastPstnExts = [];
|
||||
let pstnSort = { key: null, dir: 1 };
|
||||
|
||||
@@ -3243,6 +3312,7 @@ async function savePstnPermission(ext) {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("pstn-msg").textContent = (data.message || (data.ok ? "Saved" : "Failed")) + " (extension " + ext + ")";
|
||||
if (data.ok) markPstnDirty();
|
||||
loadPstnPermissions();
|
||||
}
|
||||
|
||||
@@ -3308,7 +3378,10 @@ document.getElementById("pd-save").addEventListener("click", async () => {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("pd-msg").textContent = data.message || (data.ok ? "Saved" : "Failed");
|
||||
if (data.ok) document.getElementById("pd-did").value = "";
|
||||
if (data.ok) {
|
||||
document.getElementById("pd-did").value = "";
|
||||
markPstnDirty();
|
||||
}
|
||||
loadPersonalDids();
|
||||
});
|
||||
|
||||
@@ -3320,6 +3393,7 @@ async function removePersonalDid(did) {
|
||||
});
|
||||
const data = await res.json();
|
||||
document.getElementById("pd-msg").textContent = data.message || (data.ok ? "Removed" : "Failed");
|
||||
if (data.ok) markPstnDirty();
|
||||
loadPersonalDids();
|
||||
}
|
||||
|
||||
@@ -3441,6 +3515,9 @@ class Handler(BaseHTTPRequestHandler):
|
||||
elif self.path == "/api/pstn-groups/apply-messaging":
|
||||
ok, message = apply_group_messaging(payload.get("name", ""), bool(payload.get("enabled", False)))
|
||||
self._json({"ok": ok, "message": message})
|
||||
elif self.path == "/api/asterisk-restart":
|
||||
ok, message = restart_asterisk_container()
|
||||
self._json({"ok": ok, "message": message})
|
||||
elif self.path == "/api/ea-devices":
|
||||
ok, result = ea_add_device(
|
||||
payload.get("name", ""), payload.get("category", ""), payload.get("extension", ""),
|
||||
|
||||
Reference in New Issue
Block a user