VoIP - Calling With a SIP URI

Back up to the VoIP Features Category
From AAISP Support Site
Revision as of 00:23, 18 August 2018 by Reedy (talk | contribs) (→‎top: clean up, typos fixed: eg → e.g.)
The printable version is no longer supported and may have rendering errors. Please update your browser bookmarks and please use the default browser print function instead.

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

Calling a SIP URI with the 'Ekiga' softphone
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, e.g. 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 (the phone you're calling from) to support SRV records.