quick_start_guide
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quick_start_guide [2024/02/11 17:22] – trmg | quick_start_guide [2024/06/16 22:26] (current) – [Pick a trunking technology (or two!)] trmg | ||
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The first step in joining SCDP is to pick a prefix. | The first step in joining SCDP is to pick a prefix. | ||
- | ====== Pick a trunking technology ====== | + | ====== Pick a trunking technology |
- | 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, | + | 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, |
- | ===== VPN tunnel required ===== | + | 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' |
==== CEMoIP ISDN PRI ==== | ==== CEMoIP ISDN PRI ==== | ||
+ | |||
+ | **VPN tunnel required** | ||
This trunking method requires a Cisco Integrated Services Router or similar that can support a Channelized T1/E1 Circuit Emulation Module like the [[https:// | This trunking method requires a Cisco Integrated Services Router or similar that can support a Channelized T1/E1 Circuit Emulation Module like the [[https:// | ||
+ | |||
+ | This will require you to choose a node to peer with. You will want to work through the config with said node operator. | ||
==== H.323 ==== | ==== H.323 ==== | ||
- | This is the preferred trunking method if you have a Cisco Integrated Services Router or other H.323 capable device in your voice network. | + | **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. | ||
+ | |||
+ | The most common H.323 gateway used among network operators is the Cisco ISR 2800/ | ||
+ | |||
+ | < | ||
+ | voice service voip | ||
+ | | ||
+ | |||
+ | voice class codec 1 | ||
+ | codec preference 10 g711ulaw | ||
+ | codec preference 20 clear-channel | ||
+ | |||
+ | voice vad-time 65536 | ||
+ | |||
+ | voice enum-match-table 1 | ||
+ | rule 10 1 /^(.*)/ /\1/ e164-scdp.vofr.net. | ||
+ | rule 20 1 / | ||
+ | |||
+ | dial-peer voice 12220 voip | ||
+ | | ||
+ | | ||
+ | | ||
+ | | ||
+ | | ||
+ | |||
+ | </ | ||
+ | |||
+ | The above assumes you are using 1-222 as the " | ||
+ | |||
+ | Once the above has been configured, you will need to deal with the toll fraud prevention feature that is enabled by default on most platforms. | ||
+ | |||
+ | If you decide to disable it, here's how you do it: | ||
+ | |||
+ | < | ||
+ | voice service voip | ||
+ | no ip address trusted authenticate | ||
+ | </ | ||
===== IAX2 ===== | ===== IAX2 ===== | ||
- | If you do not have a device that will speak H.323. | + | **VPN tunnel NOT required, but is less fun** |
+ | |||
+ | 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 " | ||
+ | |||
+ | The sample config is for Asterisk. | ||
+ | |||
+ | Add a trunk for SCDP in iax.conf | ||
+ | |||
+ | < | ||
+ | [scdp] | ||
+ | type=user | ||
+ | context=[inbound context for SCDP calls] | ||
+ | sendani=yes | ||
+ | requirecalltoken=no | ||
+ | disallow=all | ||
+ | allow=ulaw | ||
+ | allow=alaw | ||
+ | </ | ||
+ | |||
+ | 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. | ||
+ | |||
+ | < | ||
+ | [outbound-scdp] | ||
+ | |||
+ | exten => _NXXXXXX, | ||
+ | same => n, | ||
+ | same => n, | ||
+ | |||
+ | [gosub-scdp] | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | exten => s, | ||
+ | </ | ||
+ | |||
+ | NOTE: This is a work in progress. | ||
===== SIP ===== | ===== SIP ===== | ||
- | Last resort, | + | 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 ====== | ||
- | ====== Decide who you want to peer with ====== | + | Let the other participants know that you are ready to interoperate |
- | TBD |
quick_start_guide.1707672130.txt.gz · Last modified: 2024/02/11 17:22 by trmg