VoIP - Calling With a SIP URI

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From AAISP Support Site
Revision as of 12:43, 3 November 2017 by AA-Andrew (talk | contribs)
Calling a SIP URI with the 'Ekiga' softphone

Incoming calls can be made from the internet to your number by using a SIP URI such as: 

sip:number@aa.org.uk

A call to a URI like this will be delivered just like a normal incoming call.

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 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, as shown in the image here.
  • As mentioned, this does require the client to support SRV records.
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