VoIP - Calling With a SIP URI

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, 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.

Known working clients

The following SIP clients are known to work:

Softphones

Ekiga (Linux and Windows)

Known not working clients

Linphone