VoIP Phones - Asterisk: Difference between revisions

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 ==PJSIP: Trunk registration==
-Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.
+Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.<br />
+It is recommended you read the "PJSIP: NAT Issues: Keep-Alive / Anti-Idle" section below as you may wish to comment out or drastically increase the qualify_frequency line(s) if your Asterisk is not behind NAT.
 In pjsip.conf:
@@ Line 42: / Line 43: @@
     type = aor
     contact = sip:+442082881111@voiceless.aa.net.uk
+    qualify_frequency=20
+    [aaisptrunk_servera]
+    type = aor
+    contact = sip:+442082881111@a.voiceless.aa.net.uk
+    qualify_frequency=20
+    [aaisptrunk_serverb]
+    type = aor
+    contact = sip:+442082881111@b.voiceless.aa.net.uk
     qualify_frequency=20
@@ Line 58: / Line 69: @@
     direct_media = no
     rtp_symmetric = yes
-    aors = aaisptrunk
+    aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb
     outbound_auth=auth_reg_442082881111