VoIP - Calling With a SIP URI: Difference between revisions
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*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. |
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*As mentioned, this does require the client (the phone you're calling from) to support SRV records. |
*As mentioned, this does require the client (the phone you're calling from) to support SRV records. |
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==Known working clients== |
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The following SIP clients are known to work: |
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===Softphones=== |
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Ekiga (Linux and Windows) |
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===Known not working clients=== |
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Linphone |
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[[Category:VoIP Features]] |
[[Category:VoIP Features]] |
Revision as of 12:51, 11 December 2018
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