VoIP Phones - Asterisk: Difference between revisions
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Need to register to AAISP server in order to place calls Tags: Mobile edit Mobile web edit |
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==PJSIP: Trunk registration== |
==PJSIP: Trunk registration== |
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Here is an example of a working pjsip.conf setup where Asterisk will register with A&A |
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 /> |
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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. |
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 |
outbound_auth=auth_reg_442082881111 |
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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 |
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===extensions.conf=== |
===extensions.conf=== |
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