VoIP Phones - Asterisk: Difference between revisions
Appearance
Content deleted Content added
No edit summary |
|||
| Line 99: | Line 99: | ||
In this example, extn is the extension that Asterisk will pass the call to. |
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. |
Localnet should of course be set to whatever RFC1918 range you are using on your LAN. |
||
| ⚫ | |||
==Dialplan== |
==Dialplan== |
||
| Line 114: | Line 112: | ||
exten => _X.,1,Dial(SIP/snom) |
exten => _X.,1,Dial(SIP/snom) |
||
</syntaxhighlight> |
</syntaxhighlight> |
||
| ⚫ | |||
==PJSIP: Trunk registration== |
==PJSIP: Trunk registration== |
||
| Line 175: | Line 175: | ||
contact = sip:+442082881111@voiceless.aa.net.uk |
contact = sip:+442082881111@voiceless.aa.net.uk |
||
qualify_frequency=20 |
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= |
=Further Help= |
||