VoIP Phones - Asterisk: Difference between revisions
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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.
=pjsip.conf (PJSIP)=▼
==Dialplan==
exten => _X.,1,Dial(SIP/snom)
</syntaxhighlight>
▲=pjsip.conf (PJSIP)=
==PJSIP: Trunk registration==
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=
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