VoIP Phones - Asterisk: Difference between revisions

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==PJSIP: Trunk registration==

Here is an example of a working pjsip.conf setup where Asterisk will register with A&A to receive calls.<br />

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.

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outbound_auth=auth_reg_442082881111

Calls come into the context "maininbound" in extensions.conf ~~- in this example calls get sent onto extension 222 and 205 for 20 seconds and then go to voicemail.~~

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)

~~In extensions.conf you~~ can dial out via the trunk with:

You can dial out via the trunk with:

exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)