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--- quick_start_guide [2024/06/16 19:32] – trmg
+++ quick_start_guide [2024/06/16 22:26] (current) – [Pick a trunking technology (or two!)] trmg
@@ Line 7: / Line 7: @@
 ====== Pick a trunking technology (or two!) ======
-After you've picked a prefix, you will want to choose at least one trunking technology to use for SCDP. Trunking technology support varies among participants, but the following are used in varying degrees:
+After you've picked a prefix, you will want to choose at least one trunking technology to use for SCDP. Trunking technology support varies among participants, but the following are used in varying degrees.
+Note that if you decide to use CEMoIP or H.323 and would like to establish IP peering with the network, reach out to members to get that going as we're all happy to peer.  You may want to view a looking glass (such as TRMGTel's [[https://vofr.net/lg|here]]) to get a sense of the used IP space.  If you're using default 10. or 192.168. prefixes you may need to renumber.
 ==== CEMoIP ISDN PRI ====
@@ Line 21: / Line 23: @@
 **VPN tunnel recommended**
-This is the preferred trunking method if you have a Cisco Integrated Services Router or other H.323 capable device in your voice network. This is simpler than CEMoIP but is very similar signalling wise to Q.931.  However, it does require that you also establish IP peering with the network via VPN tunnel (WireGuard is most common, but other technologies can be supported if needed) and, preferably, BGP.  This is because H.323, at least with the common equipment found among the network, is not NAT friendly.  If you have a H.323 gateway directly exposed to the Internet, TRMGTel US-West has one exposed which will handle these calls.
+This is the preferred trunking method if you have a Cisco Integrated Services Router or other H.323 capable device in your voice network. This is simpler than CEMoIP but is very similar signalling wise to Q.931.  However, it may require that you also establish IP peering with the network via VPN tunnel (WireGuard is most common, but other technologies can be supported if needed) and, preferably, BGP.  This is because H.323, at least with the common equipment found among the network, is not NAT friendly.  If you have a H.323 gateway directly exposed to the Internet, TRMGTel US-West has one exposed which will handle these calls.
-The most common H.323 gateway used among network operators is the Cisco ISR 2800/2900/3800 and similar family of routers.  Below is a basic config that should get you started:
+The most common H.323 gateway used among network operators is the Cisco ISR 2800/2900/3800 and similar family of routers.  It is assumed your unit has a basic working IP configuration.  Below is a basic voice config that should get you started:
 <code>
@@ Line 65: / Line 67: @@
 If you do not have a device that will speak H.323 or do not want to dive into CEMoIP stuffs, IAX2 is the simplest way (though arguably the least "fun") to trunk with the network.
-Configuration details TBD (soon).
+The sample config is for Asterisk.  A basic understand of Asterisk is assumed.  If you are familiar with how C*NET works, this will seem familiar.
+Add a trunk for SCDP in iax.conf
+<code>
+[scdp]
+type=user
+context=[inbound context for SCDP calls]
+sendani=yes
+requirecalltoken=no
+disallow=all
+allow=ulaw
+allow=alaw
+</code>
+Build inbound dialplan code as you see fit.
+To place calls to SCDP desitnations here's a basic outbound dialplan that should work.  Node operators typically expect to receive 7 digits.
+<code>
+[outbound-scdp]
+  exten => _NXXXXXX,1,Progress()
+  same  => n,GoSub(gosub-scdp,s,1(${EXTEN}))
+  same  => n,Goto(scdp-rls-handler,s,1)
+[gosub-scdp]
+  exten => s,1,Set(result=${ENUMLOOKUP(+${ARG1},iax2,,1,e164-scdp.vofr.net)})
+  exten => s,n,GotoIf($["${result}"!=""]?dialiax)
+  exten => s,n,Progress()
+  exten => s,n,Congestion(30)
+  exten => s,n,Return()
+  exten => s,n(dialiax),Dial(IAX2/${result},120,${DIALOPTS})
+  exten => s,n,Return()
+</code>
+NOTE: This is a work in progress.  The subroutine above for ENUM lookup will only act upon the first result.  Once a good subroutine with route failover is vetted this will be updated, but this will get things going in the short term.  Suggestions always welcome.
 ===== SIP =====
-We prefer to generally stay away from SIP, but many node operators can support this if needed but it is typically last restort.  As of this writing, there are no known node-to-node SIP trunks in service thus no "quick start" instructions for this.
+We prefer to generally stay away from SIP.  Some node operators can support this if needed but it is typically last resort.  As of this writing, there are no known node-to-node SIP trunks in service thus no "quick start" instructions for this.
 ====== Announce Your Presence ======
-Let the other participants know that you are ready to interoperate with the network.  The ENUM E.164 NAPTR database will need to be updated (currently operated by TRMGTel) and node operators who are not currently utilizing the ENUM database will need to add routes manually.
+Let the other participants know that you are ready to interoperate with the network.  The ENUM E.164 NAPTR database will need to be updated (currently operated by TRMGTel) and node operators who are not currently utilizing the ENUM database may need to add routes manually.