VoIP SIP Trunks: Difference between revisions

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Revision as of 11:43, 22 June 2016 (view source) AA-Andrew (talk \| contribs) m (→‎Base/Group Numbers) ← Older edit		Revision as of 10:21, 21 March 2019 (view source) AA-James (talk \| contribs) (→‎CLI/Presentation Number) Newer edit →
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==CLI/Presentation Number== ~~For outbound calls, we will~~We allow you to present the CLI of any number ~~which~~where ~~has~~the aconfigured base number ~~that~~is ~~points~~one toyou ~~the number you're~~are using asfor authentication. We are also able to set up an additional number on request, see [[VoIP Presentation Number]]. Inbound calls can be routed to a registered SIP phone, redirected to another number or routed to your own SIP server. To help where a customer may have a block of numbers the AAISP Control Pages have a concept of grouping numbers together to use a common number as their settings. When we route calls to your own SIP server you do not need to register against our servers. You may if you wish, but we will then send INVITEs to both the registered device and to the SIP server set in the control pages. =Base/Group Numbers= If you are using the asterisk PBX, or systems based on asterisk, please see [[VoIP Phones - Asterisk]] for notes on configuration. ~~==Other VoIP Features==~~ ~~<ncl style=bullet maxdepth=5 headings=bullet headstart=2 showcats=1 showarts=1>Category:VoIP Features</ncl>~~