diff --git a/acSipBridge/astrocom.md b/acSipBridge/astrocom.md
new file mode 100644
index 0000000..e1cef4c
--- /dev/null
+++ b/acSipBridge/astrocom.md
@@ -0,0 +1,52 @@
+AstroCom integration for freePBX (Originally made for TandmX by @lachesis_._ @lachesis_._)
+1. (caveats: assumes full exchange, assumes 4 digit extensions, assumes you're okay with AstroCom users dialing anything 4-digit on your switch)
+2. Put the following in your `globals_custom.conf`:
+```
+astrocomkey=(long-key-you-got-from-admins)
+```
+3. Put the following in your `iax_custom.conf`:
+```
+[from-astrocom]
+type=user
+username=from-astrocom
+secret=(short-secret-you-sent-to-astrocom)
+auth=md5
+encryption=yes
+context=from-astrocom
+```
+4. Put the following in `extensions_custom.conf`, *change 777 to your AstroCom exchange prefix*, and uncomment one or both blocks directly below
+```
+[from-internal-additional-custom]
+; Change 777 to your exchange prefix
+; Uncomment for 7 digit dialing to AstroCom
+;exten => _XXXXXXX,1,NoOp
+;  same => n,Set(CALLERID(num)=777${CALLERID(num)})
+;  same => n,Goto(astrocom-dial,${EXTEN},1)
+;  same => n,Hangup
+
+; Uncomment for 1-300-XXX-XXXX to AstroCom
+;exten => _1300XXXXXXX,1,NoOp
+;  same => n,Set(CALLERID(num)=777${CALLERID(num)})
+;  same => n,Goto(astrocom-dial,${EXTEN:4},1)
+;  same => n,Hangup
+
+[astrocom-dial]
+exten => _X!,1,Set(number=${EXTEN})
+  same => n,Set(lookup=${CURL(https://api.astrocom.tel/api/v1/route/${astrocomkey}/${FILTER(0-9,${CALLERID(num)})}/${FILTER(0-9,${number})})})
+  same => n,GotoIf($["${lookup}"=="local"]?local:long)
+  same => n(local),Goto(astrocom-exchange,${number},1)
+  same => n(long),Dial(${lookup})
+  same => n,Hangup
+
+[from-astrocom]
+exten => _X!,1,NoOp()
+  same => n,Goto(astrocom-exchange,${EXTEN},1)
+  same => n,Hangup
+
+; Change 777 to your exchange prefix
+[astrocom-exchange]
+exten => _777XXXX,1,Goto(from-internal,${EXTEN:3},1)
+  same => n,Hangup
+```
+
+Literally picked straight from tandmx (will make our own docs eventually, but this'll get us going)
\ No newline at end of file
diff --git a/acSipBridge/extensions.conf b/acSipBridge/extensions.conf
new file mode 100644
index 0000000..84d9ce7
--- /dev/null
+++ b/acSipBridge/extensions.conf
@@ -0,0 +1,161 @@
+; =====================================================;  GLOBALS REQUIRED (globals.conf or globals_custom.conf)
+; -------------------------------------------------------------
+;  [globals]
+;  BRIDGE1_APIKEY=<routes.apiKey for bridge 1>
+;  BRIDGE1_EXCHANGE=777   ; NXX prefix — used to build the outbound ANI only,
+;                         ; NOT applied to inbound DIDs (full number passes through)
+;  BRIDGE2_APIKEY=<routes.apiKey for bridge 2>
+;  BRIDGE2_EXCHANGE=888
+;  ; ...add one pair per bridge; AstroCom SIP Bridge -- Dialplan (extensions.conf)
+; /etc/asterisk/extensions.conf
+;
+; CALL FLOWS
+; -------------------------------------------------------------
+;
+;  INBOUND  (AstroCom -> downstream PBX via PJSIP)
+;  -----------------------------------------------
+;  AstroCom delivers the full 7-digit number, e.g. 7771234.
+;  The full number is forwarded as-is to the downstream PBX as
+;  the SIP Request-URI / To: user (DID).  The PBX is responsible
+;  for routing it internally however it sees fit.
+;
+;     AstroCom --IAX2--> [bridgeN-from-astrocom]
+;                                |
+;              Pass full DID (e.g. 7771234) to PBX
+;                                |
+;                   Dial PJSIP/7771234@bridgeN-pbx
+;                                |
+;                                v
+;                        Downstream PBX
+;
+;  OUTBOUND  (downstream PBX -> AstroCom via IAX2)
+;  ------------------------------------------------
+;  The PBX dials the full 7-digit AstroCom number.
+;  The bridge prepends the local exchange so the ANI is also
+;  7 digits, then hits the AstroCom routing API:
+;
+;    GET https://api.astrocom.tel/api/v1/route/<apiKey>/<ANI>/<number>
+;
+;  The API returns one of:
+;    "local"
+;      --> same block; route to this bridge's own SIP PBX
+;    "IAX2/<auth>:<secret>@<host>:<port>/<number>"
+;      --> dial that URI verbatim
+;    "<MSG_ROUTE_ADDRESS>/4xx"
+;      --> rejected; play busy/congestion toward PBX
+;
+;     Downstream PBX --SIP--> [bridgeN-from-pbx]
+;                                     |
+;              CURL api.astrocom.tel/api/v1/route/...
+;                                     |
+;                          +----------+----------+
+;                        local               IAX2 URI
+;                          |                     |
+;            Dial PJSIP/XXXX@bridgeN-pbx   Dial ${lookup}
+;
+; ISOLATION
+; -------------------------------------------------------------
+;  * Every bridge has exactly two contexts; they share nothing.
+;  * [default] silently drops anything with no context match.
+;  * Inter-bridge calls route through AstroCom automatically --
+;    no extra config needed here.
+;
+; GLOBALS REQUIRED (globals.conf or globals_custom.conf)
+; -------------------------------------------------------------
+;  [globals]
+;  BRIDGE1_APIKEY=<routes.apiKey for bridge 1>
+;  BRIDGE1_EXCHANGE=777   ; NXX prefix (3 digits) for bridge 1
+;  BRIDGE2_APIKEY=<routes.apiKey for bridge 2>
+;  BRIDGE2_EXCHANGE=888
+;  ; ...add one pair per bridge
+; ============================================================
+
+[default]
+exten => _X!,1,Verbose(2,DROP: ${EXTEN} landed in [default] -- no matching context)
+ same  => n,Hangup(21)
+
+; ============================================================
+; --- BRIDGE 1 -----------------------------------------------
+; ============================================================
+
+; Inbound from AstroCom -> forward full DID to PBX
+; Context name == routes.auth == IAX2 section name in iax.conf
+[bridge1-from-astrocom]
+exten => _XXXXXXX,1,Verbose(2,BRIDGE1 inbound from AstroCom: ${CALLERID(all)} -> ${EXTEN})
+ same  => n,Dial(PJSIP/${EXTEN}@bridge1-pbx,30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+ same  => n(busy),Busy(5)
+ same  => n(congestion),Congestion(5)
+ same  => n(noanswer),Hangup(19)
+
+; Outbound from Bridge-1 PBX -> CURL AstroCom API -> IAX2
+[bridge1-from-pbx]
+exten => _XXXXXXX,1,Verbose(2,BRIDGE1 outbound from PBX: ${CALLERID(all)} -> ${EXTEN})
+ ; Build the 7-digit ANI: exchange prefix + bare extension from CallerID
+ same  => n,Set(ANI=${BRIDGE1_EXCHANGE}${FILTER(0-9,${CALLERID(num)})})
+ ; Ask AstroCom where to send this call
+ same  => n,Set(lookup=${CURL(https://api.astrocom.tel/api/v1/route/${BRIDGE1_APIKEY}/${ANI}/${EXTEN})})
+ same  => n,Verbose(2,BRIDGE1 AstroCom lookup: ${lookup})
+ ; "local" -- callee shares our block, route to our own SIP PBX
+ same  => n,GotoIf($["${lookup}" = "local"]?local)
+ ; Empty or error URL -- reject
+ same  => n,GotoIf($["${lookup}" = ""]?reject)
+ ; Valid IAX2 URI -- dial it verbatim
+ same  => n,Dial(${lookup},30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+ ; Local path -- strip exchange prefix, dial extension on own PBX
+ same  => n(local),Verbose(2,BRIDGE1 local call: routing ${EXTEN} to own PBX)
+ same  => n,Dial(PJSIP/${EXTEN}@bridge1-pbx,30,t)
+ same  => n,Hangup()
+ ; Error paths
+ same  => n(reject),Verbose(2,BRIDGE1 AstroCom rejected call: ${lookup})
+ same  => n,Congestion(5)
+ same  => n(busy),Busy(5)
+ same  => n(congestion),Congestion(5)
+ same  => n(noanswer),Hangup(19)
+
+; ============================================================
+; --- BRIDGE 3 (copy & uncomment for each additional bridge) -
+; ============================================================
+; Also add BRIDGE3_APIKEY and BRIDGE3_EXCHANGE to globals.conf.
+;
+;[bridge3-from-astrocom]
+;exten => _X!,1,Verbose(2,BRIDGE3 inbound: ${CALLERID(all)} -> ${EXTEN})
+; same  => n,Dial(PJSIP/${EXTEN}@bridge3-pbx,30,t)
+; same  => n,Hangup()
+;
+;[bridge3-from-pbx]
+;exten => _X!,1,Verbose(2,BRIDGE3 outbound: ${CALLERID(all)} -> ${EXTEN})
+; same  => n,Set(ANI=${BRIDGE3_EXCHANGE}${FILTER(0-9,${CALLERID(num)})})
+; same  => n,Set(lookup=${CURL(https://api.astrocom.tel/api/v1/route/${BRIDGE3_APIKEY}/${ANI}/${EXTEN})})
+; same  => n,GotoIf($["${lookup}" = "local"]?local)
+; same  => n,GotoIf($["${lookup}" = ""]?reject)
+; same  => n,Dial(${lookup},30,t)
+; same  => n,Hangup()
+; same  => n(local),Dial(PJSIP/${EXTEN}@bridge3-pbx,30,t)
+; same  => n,Hangup()
+; same  => n(reject),Congestion(5)
+
+; ============================================================
+; --- OPTIONAL: Direct SIP inter-bridge shortcut -------------
+;
+; Bridges can already call each other through AstroCom with
+; zero extra config.  If you want direct SIP routing between
+; bridges (no AstroCom hop), add a higher-priority pattern for
+; the other bridge's block BEFORE the _X! catch-all.
+;
+; Example -- Bridge-1 calling Bridge-2's 888XXXX block directly:
+;
+;[bridge1-from-pbx]
+;exten => _888XXXX,1,Verbose(2,BRIDGE1->BRIDGE2 direct: ${EXTEN})
+; same  => n,Dial(PJSIP/${EXTEN}@bridge2-pbx,30,t)
+; same  => n,Hangup()
+;
+; (The _X! rule below then handles all other numbers normally.)
+; ============================================================
diff --git a/acSipBridge/extensions.conf.bak b/acSipBridge/extensions.conf.bak
new file mode 100644
index 0000000..d7d7aea
--- /dev/null
+++ b/acSipBridge/extensions.conf.bak
@@ -0,0 +1,148 @@
+; ============================================================
+; AstroCom SIP Bridge — Dialplan (extensions.conf)
+; /etc/asterisk/extensions.conf
+;
+; CALL FLOW
+; ─────────────────────────────────────────────────────────────
+;
+;   INBOUND from AstroCom (IAX2 → SIP toward downstream PBX):
+;
+;     AstroCom  ──IAX2──▶  [bridgeN-from-astrocom]
+;                                   │
+;                           Bridge to SIP peer
+;                                   │
+;                           SIP/bridgeN-pbx/<EXTEN>
+;                                   │
+;                                   ▼
+;                          Downstream PBX / phone
+;
+;   OUTBOUND from downstream PBX (SIP → IAX2 toward AstroCom):
+;
+;     Downstream PBX  ──SIP──▶  [bridgeN-from-pbx]
+;                                       │
+;                               Bridge to IAX2 peer
+;                                       │
+;                               IAX2/bridge1-astrocom/<EXTEN>
+;                                       │
+;                                       ▼
+;                                  AstroCom
+;
+; ISOLATION
+; ─────────────────────────────────────────────────────────────
+;   • Each bridge uses its own pair of contexts
+;     (bridgeN-from-astrocom / bridgeN-from-pbx).
+;   • No cross-bridge dialplan references exist UNLESS you
+;     explicitly uncomment the inter-bridge section at the
+;     bottom.  Until then each bridge is a closed island.
+;   • The [default] context is intentionally empty / drops
+;     calls to prevent anything landing there from escaping.
+; ============================================================
+
+; ============================================================
+; Safety net — calls that arrive in no recognised context are
+; silently dropped.
+; ============================================================
+[default]
+exten => _X.,1,Verbose(2,DROP: call to ${EXTEN} landed in [default] — no context matched)
+ same  => n,Hangup(21)
+
+; ============================================================
+; ─── BRIDGE 1 ───────────────────────────────────────────────
+; ============================================================
+
+; ------ Inbound from AstroCom → forward to Bridge-1 PBX ----
+; This context name MUST equal routes.auth for this bridge.
+[bridge1-from-astrocom]
+; Accept any extension that AstroCom sends for this block.
+exten => _X.,1,Verbose(2,BRIDGE1 inbound from AstroCom: ${CALLERID(all)} → ${EXTEN})
+ same  => n,Set(CALLERID(name)=${CALLERID(name)})
+ same  => n,Dial(SIP/bridge1-pbx/${EXTEN},30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+
+ same  => n(congestion),Congestion(5)
+ same  => n(busy),Busy(5)
+ same  => n(noanswer),Hangup(19)
+
+; ------ Inbound from Bridge-1 PBX → forward to AstroCom ----
+[bridge1-from-pbx]
+exten => _X.,1,Verbose(2,BRIDGE1 outbound from PBX: ${CALLERID(all)} → ${EXTEN})
+ ; Strip any leading '9' access code the PBX may prepend, e.g. "9XXXXXXX"
+ ; Uncomment/adapt if needed:
+ ;same => n,Set(EXTEN=${EXTEN:1})
+ same  => n,Set(CALLERID(name)=${CALLERID(name)})
+ same  => n,Dial(IAX2/bridge1-astrocom/${EXTEN},30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+
+ same  => n(congestion),Congestion(5)
+ same  => n(busy),Busy(5)
+ same  => n(noanswer),Hangup(19)
+
+; ============================================================
+; ─── BRIDGE 2 ───────────────────────────────────────────────
+; ============================================================
+
+[bridge2-from-astrocom]
+exten => _X.,1,Verbose(2,BRIDGE2 inbound from AstroCom: ${CALLERID(all)} → ${EXTEN})
+ same  => n,Set(CALLERID(name)=${CALLERID(name)})
+ same  => n,Dial(SIP/bridge2-pbx/${EXTEN},30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+
+ same  => n(congestion),Congestion(5)
+ same  => n(busy),Busy(5)
+ same  => n(noanswer),Hangup(19)
+
+[bridge2-from-pbx]
+exten => _X.,1,Verbose(2,BRIDGE2 outbound from PBX: ${CALLERID(all)} → ${EXTEN})
+ same  => n,Set(CALLERID(name)=${CALLERID(name)})
+ same  => n,Dial(IAX2/bridge2-astrocom/${EXTEN},30,t)
+ same  => n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?congestion)
+ same  => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy)
+ same  => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?noanswer)
+ same  => n,Hangup()
+
+ same  => n(congestion),Congestion(5)
+ same  => n(busy),Busy(5)
+ same  => n(noanswer),Hangup(19)
+
+; ============================================================
+; ─── BRIDGE 3 (template — duplicate & uncomment) ────────────
+; ============================================================
+;[bridge3-from-astrocom]
+;exten => _X.,1,Verbose(2,BRIDGE3 inbound from AstroCom: ${CALLERID(all)} → ${EXTEN})
+; same  => n,Dial(SIP/bridge3-pbx/${EXTEN},30,t)
+; same  => n,Hangup()
+;
+;[bridge3-from-pbx]
+;exten => _X.,1,Verbose(2,BRIDGE3 outbound from PBX: ${CALLERID(all)} → ${EXTEN})
+; same  => n,Dial(IAX2/bridge3-astrocom/${EXTEN},30,t)
+; same  => n,Hangup()
+
+; ============================================================
+; ─── OPTIONAL: Inter-bridge dialling ────────────────────────
+;
+; If you want Bridge-1's PBX to be able to call Bridge-2's
+; numbers *directly* (without round-tripping through AstroCom)
+; you can include the other bridge's inbound SIP context here.
+;
+; Example: in [bridge1-from-pbx], add a pattern for Bridge-2's
+; number block *before* the catch-all _X. rule so it is
+; preferred.  Bridge-2's block in AstroCom is consulted to
+; determine the number range.
+;
+;[bridge1-from-pbx]
+;exten => _7XXXX,1,Verbose(2,BRIDGE1→BRIDGE2 direct: ${EXTEN})
+; same  => n,Dial(SIP/bridge2-pbx/${EXTEN},30,t)
+; same  => n,Hangup()
+;
+; Without uncommenting the above, bridge1 and bridge2 can still
+; reach each other — calls just travel via AstroCom as normal.
+; ============================================================
diff --git a/acSipBridge/globals.conf b/acSipBridge/globals.conf
new file mode 100644
index 0000000..e69de29
diff --git a/acSipBridge/iax.conf b/acSipBridge/iax.conf
new file mode 100644
index 0000000..0791761
--- /dev/null
+++ b/acSipBridge/iax.conf
@@ -0,0 +1,71 @@
+; ============================================================
+; AstroCom SIP Bridge — IAX2 Configuration (chan_iax2)
+; /etc/asterisk/iax.conf
+;
+; HOW ASTROCOM CALLS THIS BRIDGE
+; ──────────────────────────────
+; When AstroCom routes a call here it dials:
+;   IAX2/<routes.auth>:<routes.secret>@<routes.server>:<routes.port>/<number>
+;
+; That means:
+;   • The SECTION NAME (in brackets) IS the IAX2 username AstroCom
+;     authenticates with.  It must equal routes.auth exactly.
+;   • 'secret' must equal routes.secret exactly.
+;   • 'context' is where the call lands in extensions.conf.
+;     It must also equal routes.auth (the dialplan context is
+;     named identically for clarity).
+;   • type=user — AstroCom calls us; we never initiate an IAX2
+;     connection back to AstroCom.  Outbound calls use a
+;     dynamically-generated IAX2 URI from the routing API
+;     (handled in extensions.conf) and need no peer config here.
+;
+; NAT NOTE
+; ────────
+; Do NOT use 'register =>'.  If this server has a dynamic IP,
+; use tools/astrocom_dynamic_ip.sh (or the /api/v1/user/update
+; endpoint) to keep routes.server current in AstroCom.
+; ============================================================
+
+[general]
+bindport=4569
+bindaddr=0.0.0.0
+iaxcompat=yes
+authdebug=no
+bandwidth=high
+jitterbuffer=yes
+forcejitterbuffer=yes
+dropcount=2
+maxjitterbuffer=1000
+jittertargetextra=40
+
+; ============================================================
+; Shared user template
+; ============================================================
+[bridge-iax-tmpl](!)
+type=user
+auth=md5
+encryption=yes
+disallow=all
+allow=ulaw
+allow=alaw
+allow=g729
+qualify=yes
+
+; ============================================================
+; BRIDGE 1
+;   routes.auth   = bridge1-from-astrocom  ← section name
+;   routes.secret = BRIDGE1_IAX_SECRET
+;   routes.server = <this server's public IP>
+;   routes.port   = 4569
+; ============================================================
+[bridge1-from-astrocom](bridge-iax-tmpl)
+secret=20fbb81ba96091af1fe45858a070c7            ; Must equal routes.secret in AstroCom
+context=bridge1-from-astrocom        ; Must equal routes.auth in AstroCom
+
+
+; ============================================================
+; BRIDGE 3  (copy this block for every additional bridge)
+; ============================================================
+;[bridge3-from-astrocom](bridge-iax-tmpl)
+;secret=BRIDGE3_IAX_SECRET
+;context=bridge3-from-astrocom
diff --git a/acSipBridge/pjsip.conf b/acSipBridge/pjsip.conf
new file mode 100644
index 0000000..6c99f9b
--- /dev/null
+++ b/acSipBridge/pjsip.conf
@@ -0,0 +1,112 @@
+; ============================================================
+; AstroCom SIP Bridge — PJSIP Configuration (chan_pjsip)
+; /etc/asterisk/pjsip.conf
+;
+; Downstream PBXes REGISTER to this server.
+; Each bridge requires three stanzas sharing the same base name:
+;
+;   [bridgeN-pbx]       type=endpoint  — codecs, context, links to auth/aor
+;   [bridgeN-pbx]       type=aor       — stores the registered contact URI
+;   [bridgeN-pbx-auth]  type=auth      — username / password the PBX uses
+;
+; The PBX must be configured to register with:
+;   Registrar : sip:<this server's IP>:5060
+;   Username  : bridgeN-pbx  (must match auth.username below)
+;   Password  : the 'password' value in [bridgeN-pbx-auth]
+;
+; Outbound calls to this PBX from the dialplan:
+;   Dial(PJSIP/${EXTEN}@bridgeN-pbx)
+;   Asterisk substitutes the registered contact URI automatically.
+;
+; Disable/remove sip.conf (or set chan_sip bindport=0) so both
+; modules do not compete for port 5060.
+; ============================================================
+
+; ============================================================
+; Transport — one UDP listener for all bridges
+; ============================================================
+[bridge-transport]
+type=transport
+protocol=udp
+bind=0.0.0.0:5060
+; For NAT — uncomment and fill in:
+;local_net=192.168.0.0/16
+;external_signaling_address=PUBLIC_IP
+;external_media_address=PUBLIC_IP
+
+; ============================================================
+; Shared endpoint template
+; ============================================================
+[bridge-endpoint-tmpl](!)
+type=endpoint
+transport=bridge-transport
+disallow=all
+allow=ulaw
+allow=alaw
+allow=g729
+dtmf_mode=rfc4733
+; Force media through this server — required if either side is behind NAT
+direct_media=no
+
+; ============================================================
+; BRIDGE 1
+;   Auth username : bridge1-pbx
+;   Auth password : (set below — must match PBX registration password)
+;   Inbound calls from this PBX land in [bridge1-from-pbx] (extensions.conf)
+; ============================================================
+[bridge1-pbx](bridge-endpoint-tmpl)
+type=endpoint
+auth=bridge1-pbx-auth
+aors=bridge1-pbx
+context=bridge1-from-pbx
+
+[bridge1-pbx-auth]
+type=auth
+auth_type=userpass
+username=bridge1-pbx
+password=1242ebce8b420ee4723887ffe0a01211330851776862e6021f680f1406c94eff
+
+[bridge1-pbx]
+type=aor
+max_contacts=1
+remove_existing=yes
+
+; ============================================================
+; BRIDGE 2
+; ============================================================
+[bridge2-pbx](bridge-endpoint-tmpl)
+type=endpoint
+auth=bridge2-pbx-auth
+aors=bridge2-pbx
+context=bridge2-from-pbx
+
+[bridge2-pbx-auth]
+type=auth
+auth_type=userpass
+username=bridge2-pbx
+password=BRIDGE2_SIP_SECRET
+
+[bridge2-pbx]
+type=aor
+max_contacts=1
+remove_existing=yes
+
+; ============================================================
+; Additional bridges — copy this triple and increment N
+; ============================================================
+;[bridge3-pbx](bridge-endpoint-tmpl)
+;type=endpoint
+;auth=bridge3-pbx-auth
+;aors=bridge3-pbx
+;context=bridge3-from-pbx
+;
+;[bridge3-pbx-auth]
+;type=auth
+;auth_type=userpass
+;username=bridge3-pbx
+;password=BRIDGE3_SIP_SECRET
+;
+;[bridge3-pbx]
+;type=aor
+;max_contacts=1
+;remove_existing=yes
diff --git a/acSipBridge/sip.conf b/acSipBridge/sip.conf
new file mode 100644
index 0000000..1401c84
--- /dev/null
+++ b/acSipBridge/sip.conf
@@ -0,0 +1,74 @@
+; ============================================================
+; AstroCom SIP Bridge — SIP Configuration (chan_sip)
+; /etc/asterisk/sip.conf
+;
+; Downstream PBXes REGISTER to this server.
+; Each bridge has one [bridgeN-pbx] peer with host=dynamic.
+;
+; The PBX must be configured to register with:
+;   Registrar : <this server's IP>:5060
+;   Username  : bridgeN-pbx  (the section name below)
+;   Password  : the 'secret' value below
+;
+; 'context' is where calls FROM the PBX land (outbound to AstroCom).
+; It must match [bridgeN-from-pbx] in extensions.conf.
+; ============================================================
+
+[general]
+bindport=5060
+bindaddr=0.0.0.0
+realm=ac-sip-bridge
+srvlookup=no
+; NAT handling — set to 'force_rport,comedia' if PBX is behind NAT
+nat=no
+qualify=no
+disallow=all
+allow=ulaw
+allow=alaw
+allow=g729
+
+; ============================================================
+; Shared peer template
+; ============================================================
+[bridge-sip-tmpl](!)
+type=friend
+host=dynamic             ; IP is learned from the REGISTER request
+dtmfmode=rfc2833
+canreinvite=no
+disallow=all
+allow=ulaw
+allow=alaw
+allow=g729
+qualify=yes
+; No 'insecure=' — registration provides proper authentication.
+; Removing it means unauthenticated INVITEs from unknown IPs are rejected.
+
+; ============================================================
+; BRIDGE 1
+;   PBX registers as username 'bridge1-pbx' with the secret below.
+;   Outbound to this PBX: Dial(SIP/bridge1-pbx/<exten>)
+;     → delivered to the registered contact URI.
+; ============================================================
+[bridge1-pbx](bridge-sip-tmpl)
+secret=1242ebce8b420ee4723887ffe0a01211330851776862e6021f680f1406c94eff
+; If the PBX must register with a different username, set it here:
+;username=bridge1
+fromuser=bridge1-pbx     ; identity used in outbound SIP From: headers
+context=bridge1-from-pbx
+
+; ============================================================
+; BRIDGE 2
+;   PBX registers as username 'bridge2-pbx' with the secret below.
+; ============================================================
+[bridge2-pbx](bridge-sip-tmpl)
+secret=BRIDGE2_SIP_SECRET
+fromuser=bridge2-pbx
+context=bridge2-from-pbx
+
+; ============================================================
+; Additional bridges — copy this block and increment N
+; ============================================================
+;[bridge3-pbx](bridge-sip-tmpl)
+;secret=BRIDGE3_SIP_SECRET
+;fromuser=bridge3-pbx
+;context=bridge3-from-pbx
diff --git a/acSipBridge/theory.md b/acSipBridge/theory.md
new file mode 100644
index 0000000..2f1aaf1
--- /dev/null
+++ b/acSipBridge/theory.md
@@ -0,0 +1,157 @@
+# 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)
+
+1. AstroCom looks up the dialled number in its `routes` table and finds
+   the bridge server's IP (`routes.server`) and port (`routes.port`).
+2. 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`).
+3. 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)
+
+1. The downstream PBX has registered to this server as `bridge1-pbx`.
+   It sends a SIP INVITE for the target number.
+2. `sip.conf` authenticates the REGISTER/INVITE against `secret` and
+   lands the call in context `bridge1-from-pbx`.
+3. The dialplan CURLs the AstroCom routing API:
+   `GET /api/v1/route/<apiKey>/<ANI>/<number>`
+4. 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
+
+1. **`iax.conf`** — Copy the `[bridge1-from-astrocom]` block, rename it to
+   `[bridgeN-from-astrocom]`, and set a new `secret` and `context`.
+
+2. **`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]`
+
+3. **`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`.
+
+4. **AstroCom** — Create the route via the admin panel or API with
+   `auth = bridgeN-from-astrocom` and the matching `secret`.
+
+5. 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.