Differences

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--- using_freepbx_with_sdf_voip [2023/07/03 20:08] – [Using FreePBX with SDF VOIP] hc9
+++ using_freepbx_with_sdf_voip [2024/07/27 05:08] (current) – [Procedure] drelcott
@@ Line 13: / Line 13: @@
   - Go to //Connectivity// » //Trunks//
-  - Select //Add Trunk// and choose //Add SIP (chan_sip) trunk//
+  - Select //Add Trunk// and choose //Add SIP (chan_pjsip) trunk//*
   - Set //Trunk Name// to ''sdf''
   - Set //Outbound Caller ID// to ''<$EXTENSION@sip.sdf.org>''
   - Set //CID Options// to //Force Trunk CID//
-  - Go to the //sip Settings// tab
+  - Additional configuration for pjsip:
-  - Set //Trunk Name// to ''SDF''
+    - Go to the //pjsip Settings// tab
-  - Set peer details to:<code>
+      - Under the //General// tab fill out the following
+        - Username: ''$EXTENSION''
+        - Secret: ''$PASSWORD''
+        - Authentication: Outbound
+        - Registration: Send
+        - SIP Server: ''sip.sdf.org''
+        - SIP Port: ''5060''
+        - Context: ''from-trunk''
+        - Transport: ''0.0.0.0-udp''
+      - Under the //Advanced// tab, most of the defaults should work okay, but make sure the following are set:
+        - DTMF Mode: Auto
+        - From Domain: ''sip.sdf.org''
+        - From User: ''$EXTENSION''
+        - Direct Media: No
+     - Click //Submit// and then //Apply Config//
+  - Additional configuration for sip (//legacy//):
+     - Go to the //sip Settings// tab
+       - Set //Trunk Name// to ''SDF''
+       - Set peer details to:<code>
 host=sip.sdf.org
 username=$EXTENSION
@@ Line 36: / Line 54: @@
 allow=ulaw
 </code>
-  - Go to the //Incoming// tab
+       - Go to the //Incoming// tab
-  - Set //Register String// to ''$EXTENSION:$PASSWORD@sip.sdf.org''
+       - Set //Register String// to ''$EXTENSION:$PASSWORD@sip.sdf.org''
-  - Click //Submit//
+       - Click //Submit//
-You can now define Inbound and Outbound routes using this new trunk. To test inbound calls, you can queue a test call through ''maint''.
+You can now define Inbound and Outbound routes using this new trunk. You may refer to FreePBX documentation on how to set up outbound and inbound routes. To test inbound calls, you can queue a test call through ''maint''.
+* **NOTE:** Older versions of FreePBX may still allow for chan_sip trunk settings, but this is deprecated and chan_pjsip is now the preferred configuration.
 |If you have an incoming DID through SDF, just a note that the SDF's system does not pass ''FROM_DID=''.|