VoIP Phones - Asterisk: Difference between revisions
Content deleted Content added
Need to register to AAISP server in order to place calls 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: |
In pjsip.conf: |
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[ |
[reg_441234567890] |
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type = registration |
type = registration |
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retry_interval = 20 |
retry_interval = 20 |
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contact_user = maininbound |
contact_user = maininbound |
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expiration = 120 |
expiration = 120 |
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outbound_auth = |
outbound_auth = auth_reg_441234567890 |
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client_uri = sip:+ |
client_uri = sip:+441234567890@voiceless.aa.net.uk |
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server_uri = sip:voiceless.aa.net.uk |
server_uri = sip:voiceless.aa.net.uk |
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[ |
[auth_reg_441234567890] |
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type = auth |
type = auth |
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password = |
password = SecretPasswordGoesHere |
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username = + |
username = +441234567890 |
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[aaisptrunk] |
[aaisptrunk] |
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type = aor |
type = aor |
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contact = sip:+ |
contact = sip:+441234567890@voiceless.aa.net.uk |
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qualify_frequency=20 |
qualify_frequency=20 |
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[aaisptrunk_servera] |
[aaisptrunk_servera] |
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type = aor |
type = aor |
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contact = sip:+ |
contact = sip:+441234567890@a.voiceless.aa.net.uk |
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qualify_frequency=20 |
qualify_frequency=20 |
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[aaisptrunk_serverb] |
[aaisptrunk_serverb] |
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type = aor |
type = aor |
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contact = sip:+ |
contact = sip:+441234567890@b.voiceless.aa.net.uk |
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qualify_frequency=20 |
qualify_frequency=20 |
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disallow = all |
disallow = all |
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allow = alaw |
allow = alaw |
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allow = ulaw |
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direct_media = no |
direct_media = no |
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rtp_symmetric = yes |
rtp_symmetric = yes |
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aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb |
aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb |
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outbound_auth= |
outbound_auth=auth_reg_441234567890 |
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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 |
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 |
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exten = maininbound,n,Voicemail(222@default,us) |
exten = maininbound,n,Voicemail(222@default,us) |
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You can dial out via the trunk with: |
You can dial out via the trunk with (probably in a context like "from-internal"): |
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[mainoutbound] |
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exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,) |
exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,) |
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==PJSIP: Trunk without registration== |
==PJSIP: Trunk without registration== |
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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. |
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Then set the AAISP control panel to point to your server by hostname or IP address:<br /> |
Then set the AAISP control panel to point to your server by hostname or IP address:<br /> |
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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. |
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[[File:Asterisk pjsip noregistration.png|border]] |
[[File:Asterisk pjsip noregistration.png|border]] |
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exten => _X.,1,Dial(SIP/voiceless-out/${EXTEN}) |
exten => _X.,1,Dial(SIP/voiceless-out/${EXTEN}) |
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</syntaxhighlight> |
</syntaxhighlight> |
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For inbound calls (assuming you're routing |
For inbound calls (assuming you're routing calls to a context named "snom"): |
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<syntaxhighlight lang="ini"> |
<syntaxhighlight lang="ini"> |
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[voiceless-in] |
[voiceless-in] |
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