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VoIP Phones - Asterisk: Difference between revisions
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Don't feel comfortable with live number in example. No point in allowing ulaw because AAISP don't use it
m (It's not that the phone is a Snom, it's the context which it's in) Tags: Mobile edit Mobile web edit |
m (Don't feel comfortable with live number in example. No point in allowing ulaw because AAISP don't use it) |
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In pjsip.conf:
[
type = registration
retry_interval = 20
contact_user = maininbound
expiration = 120
outbound_auth =
client_uri = sip:+
server_uri = sip:voiceless.aa.net.uk
[
type = auth
password =
username = +
[aaisptrunk]
type = aor
contact = sip:+
qualify_frequency=20
[aaisptrunk_servera]
type = aor
contact = sip:+
qualify_frequency=20
[aaisptrunk_serverb]
type = aor
contact = sip:+
qualify_frequency=20
disallow = all
allow = alaw
direct_media = no
rtp_symmetric = yes
aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb
outbound_auth=
The "contact_user" option in the registration section sets the context for incoming calls to Asterisk, in this example calls come into the context "maininbound" in extensions.conf
exten = maininbound,n,Voicemail(222@default,us)
You can dial out via the trunk with (probably in a context like "from-internal"):
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,)
Then set the AAISP control panel to point to your server by hostname or IP address:<br />
Outgoing calls require registration, and you'll automatically receive incoming calls to registered "phones". If you register Asterisk and have calls sent directly to Asterisk you'll receive 2 copies of each call.
[[File:Asterisk pjsip noregistration.png|border]]
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