VoIP Phones - Asterisk: Difference between revisions

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 = Configuration =
+Asterisk has two methods to configure SIP connections. The legacy "sip.conf" (SIP) and the more modern "pjsip.conf" (PJSIP).
+Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues. It isn't a good idea to have an installation that mixes sip.conf with pjsip.conf.
+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.
 == Incoming Calls ==
 === User Section ===
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 ==Note: Order of sip.conf is important==
 In sip.conf, ensure that your incoming config is before the config for the outgoing.
-==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
 ==Note: Asterisk and IPv6 SLAAC addresses==
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 Localnet should of course be set to whatever RFC1918 range you are using on your LAN.
+=PJSIP=
 ==Dialplan==
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 </syntaxhighlight>
+==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:
+ Dial(PJSIP/0800500005@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
 =Further Help=