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~~ in the same way as a SIP phone does.<br />

Here is an example of a working pjsip.conf setup where Asterisk will register with A&A in the same way as a SIP phone does in order to make outgoing calls. Incoming calls are sent to all registered SIP "phones".<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:

[~~reg_442082881111~~]

[reg_441234567890]

type = registration

retry_interval = 20

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contact_user = maininbound

expiration = 120

outbound_auth = ~~auth_reg_442082881111~~

outbound_auth = auth_reg_441234567890

client_uri = sip:+~~442082881111~~@voiceless.aa.net.uk

client_uri = sip:+441234567890@voiceless.aa.net.uk

server_uri = sip:voiceless.aa.net.uk

[~~auth_reg_442082881111~~]

[auth_reg_441234567890]

type = auth

password = ~~NotRealPasswordHere~~

password = SecretPasswordGoesHere

username = +~~442082881111~~

username = +441234567890

[aaisptrunk]

type = aor

contact = sip:+~~442082881111~~@voiceless.aa.net.uk

contact = sip:+441234567890@voiceless.aa.net.uk

qualify_frequency=20

[aaisptrunk_servera]

type = aor

contact = sip:+~~442082881111~~@a.voiceless.aa.net.uk

contact = sip:+441234567890@a.voiceless.aa.net.uk

qualify_frequency=20

[aaisptrunk_serverb]

type = aor

contact = sip:+~~442082881111~~@b.voiceless.aa.net.uk

contact = sip:+441234567890@b.voiceless.aa.net.uk

qualify_frequency=20

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disallow = all

allow = alaw

allow = ulaw

direct_media = no

rtp_symmetric = yes

aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb

outbound_auth=~~auth_reg_442082881111~~

outbound_auth=auth_reg_441234567890

~~Calls~~ come into the context "maininbound" in extensions.conf

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

===extensions.conf===

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exten = maininbound,n,Voicemail(222@default,us)

You can dial out via the trunk with:

You can dial out via the trunk with (probably in a context like "from-internal"):

[mainoutbound]

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

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==PJSIP: Trunk without registration==

As an ~~alternative~~ ~~to registering~~ Asterisk ~~like~~ it ~~were~~ ~~a SIP phone~~, you can have A&A send incoming calls to Asterisk. Use the above example but do not include the top section for "[reg_442082881111]".

If you don't need Asterisk to make outgoing calls, you can have A&A send incoming calls directly to Asterisk. Use the above example but do not include the top section for "[reg_442082881111]" and the 'outbound_auth' item in the aaisptrunk endpoint.

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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exten => _X.,1,Dial(SIP/voiceless-out/${EXTEN})

</syntaxhighlight>

For inbound calls (assuming you're routing ~~call~~ to a ~~registered~~ ~~Snom~~):

For inbound calls (assuming you're routing calls to a context named "snom"):

[voiceless-in]