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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.
=sip.conf (SIP)=
== Incoming Calls ==
=== User Section ===
In this example, extn is the extension that Asterisk will pass the call to.
Localnet should of course be set to whatever RFC1918 range you are using on your LAN.
==Dialplan==
exten => _X.,1,Dial(SIP/snom)
</syntaxhighlight>
=pjsip.conf (PJSIP)=
==PJSIP: Trunk registration==
In extensions.conf you can dial out via the trunk with:
exten => _X.,1,Dial(PJSIP/
==PJSIP: Keep-Alive / Anti-Idle==
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.
=Further Help=
Customers using Asterisk and AAISP have created a website and IRC channel especially for this!
*http://www.aa-asterisk.org.uk/ [ Dead link @ Dec 2020 ]
*irc://
=Firewall & Security=
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