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VoIP Phones - Asterisk: Difference between revisions
→PJSIP: Trunk registration
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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 />
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.
In pjsip.conf:
type = aor
contact = sip:+442082881111@voiceless.aa.net.uk
qualify_frequency=20
[aaisptrunk_servera]
type = aor
contact = sip:+442082881111@a.voiceless.aa.net.uk
qualify_frequency=20
[aaisptrunk_serverb]
type = aor
contact = sip:+442082881111@b.voiceless.aa.net.uk
qualify_frequency=20
direct_media = no
rtp_symmetric = yes
aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb
outbound_auth=auth_reg_442082881111
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