VoIP Phones - Asterisk: Difference between revisions

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 When reading the instructions below be aware which are for sip.conf and which are for pjsip.conf. PJSIP examples are below the SIP examples on this page.
+=pjsip.conf (PJSIP)=
+==PJSIP: Trunk registration==
+Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.
+In pjsip.conf:
+    [reg_442082881111]
+    type = registration
+    retry_interval = 20
+    fatal_retry_interval = 20
+    forbidden_retry_interval = 20
+    max_retries = 9999
+    auth_rejection_permanent = no
+    contact_user = maininbound
+    expiration = 120
+    outbound_auth = auth_reg_442082881111
+    client_uri = sip:+442082881111@voiceless.aa.net.uk
+    server_uri = sip:voiceless.aa.net.uk
+    [auth_reg_442082881111]
+    type = auth
+    password = BusinessPaidGrewCome
+    username = +442082881111
+    [aaisptrunk]
+    type = aor
+    contact = sip:+442082881111@voiceless.aa.net.uk
+    qualify_frequency=20
+    [aaisptrunk]
+    type = identify
+    endpoint = aaisptrunk
+    match = voiceless.aa.net.uk
+    [aaisptrunk]
+    type = endpoint
+    context = maininbound
+    dtmf_mode = rfc4733
+    disallow = all
+    allow = alaw
+    allow = ulaw
+    direct_media = no
+    aors = aaisptrunk
+    outbound_auth=auth_reg_442082881111
+Calls come into the context "maininbound" in extensions.conf - in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.
+ [maininbound]
+ exten = maininbound,1,Dial(PJSIP/222&PJSIP/205,20)
+ exten = maininbound,n,Voicemail(222@default,us)
+In extensions.conf you can dial out via the trunk with:
+ exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
+==PJSIP: Keep-Alive / Anti-Idle==
+If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.
+Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds.
+An example of the aor section follows:
+ [aaisptrunk]
+ type = aor
+ contact = sip:+442082881111@voiceless.aa.net.uk
+ qualify_frequency=20
+==Status and Commands==
+A good command within the asterisk software is the show registration command:
+ asterisk*CLI> pjsip show registrations
+  <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
+ ==========================================================================================
+  reg_442082881111/sip:voiceless.aa.net.uk                auth_reg_442082881111  Registered
+ Objects found: 1
+In this example it shows that the Asterisk server is successfully registered with the Andrews & Arnold SIP server.
 =sip.conf (SIP)=
@@ Line 114: / Line 189: @@
 exten => _X.,1,Dial(SIP/snom)
 </syntaxhighlight>
-=pjsip.conf (PJSIP)=
-==PJSIP: Trunk registration==
-Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.
-In pjsip.conf:
-    [reg_442082881111]
-    type = registration
-    retry_interval = 20
-    fatal_retry_interval = 20
-    forbidden_retry_interval = 20
-    max_retries = 9999
-    auth_rejection_permanent = no
-    contact_user = maininbound
-    expiration = 120
-    outbound_auth = auth_reg_442082881111
-    client_uri = sip:+442082881111@voiceless.aa.net.uk
-    server_uri = sip:voiceless.aa.net.uk
-    [auth_reg_442082881111]
-    type = auth
-    password = BusinessPaidGrewCome
-    username = +442082881111
-    [aaisptrunk]
-    type = aor
-    contact = sip:+442082881111@voiceless.aa.net.uk
-    qualify_frequency=20
-    [aaisptrunk]
-    type = identify
-    endpoint = aaisptrunk
-    match = voiceless.aa.net.uk
-    [aaisptrunk]
-    type = endpoint
-    context = maininbound
-    dtmf_mode = rfc4733
-    disallow = all
-    allow = alaw
-    allow = ulaw
-    direct_media = no
-    aors = aaisptrunk
-    outbound_auth=auth_reg_442082881111
-Calls come into the context "maininbound" in extensions.conf - in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.
- [maininbound]
- exten = maininbound,1,Dial(PJSIP/222&PJSIP/205,20)
- exten = maininbound,n,Voicemail(222@default,us)
-In extensions.conf you can dial out via the trunk with:
- exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
-==PJSIP: Keep-Alive / Anti-Idle==
-If you are using a firewall or NAT router with short timeouts on UDP sessions you can force packets to be sent over the connection to keep it alive.
-Set qualify_frequency in the aor section; This triggers an OPTIONS message every X (as set) seconds.
-An example of the aor section follows:
- [aaisptrunk]
- type = aor
- contact = sip:+442082881111@voiceless.aa.net.uk
- qualify_frequency=20
-==Status and Commands==
-A good command within the asterisk software is the show registration command:
- asterisk*CLI> pjsip show registrations
-  <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
- ==========================================================================================
-  reg_442082881111/sip:voiceless.aa.net.uk                auth_reg_442082881111  Registered
- Objects found: 1
-In this example it shows that the Asterisk server is successfully registered with the Andrews & Arnold SIP server.
 =Firewall & Security=