VoIP - Calling With a SIP URI: Difference between revisions

Revision as of 12:43, 3 November 2017

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.

@@ Line 10: / Line 10: @@
  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.
+*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.
-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.
+*As mentioned, this does require the client to support SRV records.
-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.
 [[Category:VoIP Features]]