Project:VOIP: Difference between revisions

Project:VOIP (view source)

757 bytes added , 26 January 2019

242

edits

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 ===PJSIP===
-The chan_sip module will be going EOL in the not so distant future, so we're now using chan_pjsip - a newer, shinier, and more RFC compliant SIP stack. The syntax is significantly different, but still works much the same way.
+The SIP stack's core functionality, as well as endpoint definitions for endpoints such as phones and VoIP providers are managed by the chan_pjsip module. The syntax is somewhat different to Asterisks former chan_sip module, so ensure you're using the correct documentation when making changes.
 {| class="wikitable"
@@ Line 51: / Line 50: @@
 | pjsip_extensions.conf
 | Defines "extensions", I.E Phones registering to the PBX
+|}
+===Extensions===
+Call routing and features are configured in the extensions files.
+{| class="wikitable"
+! Filename
+! Description
+|-
+| extensions.conf
+| The default extensions.conf template that ships with Asterisk. You shouldn't need to edit this file too much.
+|-
+| extensions_globals.conf
+| Defines various global variables such as ring groups, to make editing them easier.
+|-
+| extensions_internal.conf
+| Internal calls, i.e those originating from the various phones and ATAs in the Hackspace end up here.
+|-
+| extensions_external.conf
+| External calls, i.e those originating from the PSTN end up here.
+|-
+| extensions_utils.conf
+| Various utilities and functions are defined here.
 |}