VoIP Phones - Asterisk: Difference between revisions

VoIP Phones - Asterisk (view source)

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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== In extensions.conf you can dial out via the trunk with: exten => _X.,1,Dial(PJSIP/~~0800500005~~${EXTEN}@aaisptrunk,,) ==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: asteriskCLI> 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://zirc.jeaachat.net/a&a-asterisk =Firewall & Security=