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Difference between revisions of "VoIP - Calling With a SIP URI"

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*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.
 
*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.
 
*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.
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*As mentioned, this does require the client (the phone you're calling from) to support SRV records.
   
 
[[Category:VoIP Features]]
 
[[Category:VoIP Features]]

Revision as of 07:52, 24 May 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.

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 (the phone you're calling from) to support SRV records.