Differences

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--- using_freepbx_with_sdf_voip [2021/03/20 19:23] – [Procedure] hc9
+++ using_freepbx_with_sdf_voip [2022/02/25 05:00] – hc9
@@ Line 12: / Line 12: @@
 In the admin interface follow these steps, replacing ''$EXTENSION'' and ''$PASSWORD'' with your extension number and password respectively:
-|**1**|Go to //Connectivity// » //Trunks//|
+  - Go to //Connectivity// » //Trunks//
-|**2**|Select //Add Trunk// and choose //Add SIP (chan_sip) trunk//|
+  - Select //Add Trunk// and choose //Add SIP (chan_sip) trunk//
-|**3**|Set //Trunk Name// to ''sdf''|
+  - Set //Trunk Name// to ''sdf''
-|**4**|Set //Outbound Caller ID// to ''<$EXTENSION@sip.sdf.org>''|
+  - Set //Outbound Caller ID// to ''<$EXTENSION@sip.sdf.org>''
-|**5**|Set //CID Options// to //Force Trunk CID//|
+  - Set //CID Options// to //Force Trunk CID//
-|**6**|Go to the //sip Settings// tab|
+  - Go to the //sip Settings// tab
-|**7**|Set //Trunk Name// to ''SDF''|
+  - Set //Trunk Name// to ''SDF''
-|**8**|Set peer details to:   host=sip.sdf.org|
+  - Set peer details to:<code>
+host=sip.sdf.org
-<code>
+username=$EXTENSION
+secret=$PASSWORD
-  username=$EXTENSION
+type=friend
-  secret=$PASSWORD
+context=from-trunk
-  type=friend
+insecure=port,invite
-  context=from-trunk
+trustrpid=yes
-  insecure=port,invite
+sendrpid=yes
-  trustrpid=yes
+directmedia=no
-  sendrpid=yes
+qualify=yes
-  directmedia=no
+keepalive=45
-  qualify=yes
+nat=yes
-  keepalive=45
+dtmfmode=rfc2833
-  nat=yes
+disallow=all
-  dtmfmode=rfc2833
+allow=ulaw
-  disallow=all
-  allow=ulaw
 </code>
+  - Go to the //Incoming// tab
+  - Set //Register String// to ''$EXTENSION:$PASSWORD@sip.sdf.org''
+  - Click //Submit//
-|**9**|Go to the //Incoming// tab|
+You can now define Inbound and Outbound routes using this new trunk. To test inbound calls, you can queue a test call through ''maint''.
-|**10**|Set //Register String// to ''$EXTENSION:$PASSWORD@sip.sdf.org''|
-|**11**|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''.
+|If you have an incoming DID through SDF, just a note that the SDF's system does not pass ''FROM_DID=''.|
+|If you would like to route your call through a specific incoming route by DID, then add the DID after the registration string.|
+|For example: ''$EXTENSION:$PASSWORD@sip.sdf.org/$DID'' You can add the 11 or 10 digit number, just make sure it matches exactly to what you want in the incoming route.|
 $Id: sdf_voip_freepbx.html,v 1.2 2017/11/07 12:49:54 irl Exp $ [[http://sdf.org/?tutorials/sdf_voip_freepbx|Using FreePBX with SDF VOIP]] - traditional link (using [[wp>Revision_Control_System|RCS]])