AstroCom SIP Bridge — Design Notes

This directory contains a set of Asterisk configuration files that act as a protocol bridge between IAX2 (used by AstroCom) and SIP (used by downstream, non-Asterisk phone systems such as FreeSWITCH, 3CX, Cisco UCM, or any generic SIP PBX).

Files

File	Purpose
`iax.conf`	IAX2 peers — one per bridge, pointing at AstroCom
`pjsip.conf`	PJSIP endpoints — one triple (endpoint/auth/aor) per bridge
`extensions.conf`	Dialplan — contexts that wire IAX↔PJSIP for each bridge

Architecture

                         ┌─────────────────────────────────┐
                         │       AstroCom Platform          │
                         │  (IAX2 server, routes DB)        │
                         └──────────┬──────────────────────-┘
                                    │  IAX2 (port 4569)
                    ┌───────────────┼────────────────┐
                    │               │                │
              IAX2 peer       IAX2 peer         IAX2 peer
            bridge1-astrocom  bridge2-astrocom  bridgeN-astrocom
                    │               │                │
            ┌───────┴───────────────┴────────────────┴──────┐
            │               Bridge Server                    │
            │           (this Asterisk instance)             │
            │                                                │
            │  [bridge1-from-astrocom]  [bridge1-from-pbx]  │
            │  [bridge2-from-astrocom]  [bridge2-from-pbx]  │
            │  [bridgeN-from-astrocom]  [bridgeN-from-pbx]  │
            └───────┬───────────────┬────────────────┬──────┘
                    │               │                │
              SIP peer        SIP peer          SIP peer
             bridge1-pbx     bridge2-pbx       bridgeN-pbx
                    │               │                │
                    ▼               ▼                ▼
              PBX / System 1  PBX / System 2  PBX / System N

Inbound call (AstroCom → PBX)

AstroCom looks up the dialled number in its routes table and finds the bridge server's IP (routes.server) and port (routes.port).
AstroCom dials over IAX2 using the shared secret (routes.secret) and delivers the call to the context named in routes.auth (e.g. bridge1-from-astrocom).
The bridge's dialplan receives the call in [bridge1-from-astrocom] and dials SIP/bridge1-pbx/<EXTEN>. Because the PBX has already registered (host=dynamic), Asterisk knows the current contact URI and sends the INVITE directly to it.

Outbound call (PBX → AstroCom)

The downstream PBX has registered to this server as bridge1-pbx. It sends a SIP INVITE for the target number.
sip.conf authenticates the REGISTER/INVITE against secret and lands the call in context bridge1-from-pbx.
The dialplan CURLs the AstroCom routing API: GET /api/v1/route/<apiKey>/<ANI>/<number>
The API returns either "local" (callee is on the same PBX — route back via SIP/bridge1-pbx/<exten>) or a full IAX2/<auth>:<secret>@<host>:<port>/<number> URI which is dialled verbatim.

AstroCom Database Requirements

For each bridge you must create one row in the routes table:

Column	Value
`server`	Bridge server's public IP
`port`	`4569` (IAX2)
`auth`	`bridgeN-from-astrocom` (must match context name)
`secret`	The shared IAX2 secret (`BRIDGEN_IAX_SECRET`)
`block_start`	First number in this bridge's number block
`block_length`	Size of the block (default 9999)
`apiKey`	Auto-generated by AstroCom on creation

Isolation Model

Each bridge is completely isolated by default:

IAX2 peers use separate contexts (bridge1-from-astrocom, etc.)
PJSIP endpoints use separate contexts (bridge1-from-pbx, etc.)
No context includes or jumps to another bridge's context
The [default] context drops all unmatched calls

Calling between bridges

Calls between Bridge-1 and Bridge-2 work through AstroCom's normal routing — Bridge-1's PBX dials a number, it travels to AstroCom via IAX2, AstroCom routes it to Bridge-2's context, and it arrives at Bridge-2's PBX via PJSIP. No special config is needed for this.

If you want direct PJSIP bridging (bypassing AstroCom for inter-bridge calls), see the commented-out section at the bottom of extensions.conf.

Adding a New Bridge

iax.conf — Copy the [bridge1-from-astrocom] block, rename it to [bridgeN-from-astrocom], and set a new secret and context.
pjsip.conf — Copy the three stanzas for bridge1-pbx, rename them to bridgeN-pbx / bridgeN-pbx-auth, set a new password, and update context. The downstream PBX must be configured to register with:
- Registrar: sip:<bridge server IP>:5060
- Username: bridgeN-pbx
- Password: the password value in [bridgeN-pbx-auth]
extensions.conf — Copy both the [bridge1-from-astrocom] and [bridge1-from-pbx] context blocks, rename them to [bridgeN-from-astrocom] / [bridgeN-from-pbx], and update the Dial() targets and global variable names. Add BRIDGEN_APIKEY and BRIDGEN_EXCHANGE to globals.conf.
AstroCom — Create the route via the admin panel or API with auth = bridgeN-from-astrocom and the matching secret.

Reload Asterisk:

asterisk -rx "pjsip reload"
asterisk -rx "iax2 reload"
asterisk -rx "dialplan reload"

Notes

Codecs: ulaw, alaw, and g729 are negotiated; adjust per your network and licensing constraints.
NAT: If a downstream PBX is behind NAT, set direct_media=no (already in the template) and add local_net / external_signaling_address / external_media_address to [bridge-transport] in pjsip.conf.
No register => for IAX2: IP updates use tools/astrocom_dynamic_ip.sh or the /api/v1/user/update API endpoint — not IAX2 registration.
Security: Each bridge's IAX2 secret should be a long random string (AstroCom generates one via crypto.randomBytes(15).toString('hex') if none is specified during route creation).
sip.conf: The old chan_sip config is superseded by pjsip.conf. Ensure chan_sip is not loaded (noload => chan_sip.so in modules.conf) to avoid both modules fighting over port 5060.

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