VoIP - Calling With a SIP URI: Difference between revisions

Revision as of 23:26, 11 December 2018

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. 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:

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.

The following SIP clients are known to work:

@@ Line 5: / Line 5: @@
  sip:number@aa.org.uk
-A call to a URI like this will be delivered just like a normal incoming call.
+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:
@@ Line 19: / Line 19: @@
 ===Softphones===
-Ekiga (Linux and Windows)
+* Ekiga (Linux)
+* Linphone 4.0.1 (Android)
+* Zoiper (Android)
-===Known not working clients===
+==Known not-working clients==
-Linphone
+* Linphone 3.6.1 (Linux)
 [[Category:VoIP Features]]