Difference between revisions of "VoIP - Calling With a SIP URI"

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(Created page with "__NOTOC__<indicator name="Configuring">link=:Category:VoIP Features|30px|Back up to the VoIP Features Category</indicator> Incoming calls can be made fr...")
 
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__NOTOC__<indicator name="Configuring">[[File:Menu-cog.svg|link=:Category:VoIP Features|30px|Back up to the VoIP Features Category]]</indicator>
 
Incoming calls can be made from the internet to sip:number@aa.org.uk and delivered just like a normal incoming call. The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted.
 
The format of your number should be in +44 format, eg to call AAISP support you'd use: sip:+443333400999@aa.org.uk
 
You do need to use the hostname ''aa.org.uk'' as we use SRV records to direct the call, so using an IP address instead is not supported.
 
The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted.
 
So, for example, without any configuration or account details a softphone may call your A&A provided VoIP number.
 
[[File:VOIP-sipURIcall.png|thumb|Calling a SIP URI]]
 
[[Category:VoIP Features]]

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