Back up to the VoIP Features Category

VoIP - Calling With a SIP URI

From AAISP Support Site
Revision as of 00:26, 12 December 2018 by Adsb (talk | contribs) (Add some tested clients)
Jump to navigation Jump to search
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:  

A call to a URI like this will be delivered just like a normal incoming call. Except the caller won't incur any call charges.

The format of your number should be in +44 format, e.g. to call AAISP support you'd use:
  • You do need to use the hostname 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:


  • Ekiga (Linux)
  • Linphone 4.0.1 (Android)
  • Zoiper (Android)

Known not-working clients

  • Linphone 3.6.1 (Linux)