editor
520
edits
VoIP Phones - Asterisk: Difference between revisions
m
Put the extensions.conf entries under a header
m (Use UK call progress tones) |
m (Put the extensions.conf entries under a header) |
||
|
==PJSIP: Trunk registration==
Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls in the same way as a SIP phone does.<br />
It is recommended you read the "PJSIP: NAT Issues: Keep-Alive / Anti-Idle" section below as you may wish to comment out or drastically increase the qualify_frequency line(s) if your Asterisk is not behind NAT.
outbound_auth=auth_reg_442082881111
Calls come into the context "maininbound" in extensions.conf
===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)
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
| |||