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

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Get NetworkManager to takeover control of /etc/resolv.conf
m (Twinkle works, if you disable systemd-resolved)
m (Get NetworkManager to takeover control of /etc/resolv.conf)
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[[File:VOIP-sipURIcall.png|thumb|Calling a SIP URI with the 'Ekiga' softphone]]
With SIP, it's possible to make data calls over the Internet to SIP accounts by using URIs of the form ''sip:user@domain''. The ''user'' field can be a name or a phone number. Such calls don't use the normal phone networks, and aren't charged as phone calls (although you may have to pay for the data used, like any other Internet traffic).
Incoming calls can be made from the internet to your number by using a SIP URI such as:
For example:
* AAISP sales can be contacted at sip:sales@aa.net.uk
* AAISP support can be contacted at sip:support@aa.net.uk
* AAISP accounts an be contacted at sip:accounts@aa.net.uk
* the acclaimed 'Lenny 'service is supposedly available at sip:13475147296@in.callcentric.com
Incoming calls can be made from the internetInternet to youran AAISP number by using a SIP URI such as:
A call to a URI like this will be delivered just like a normal incoming call. - Exceptexcept the caller won't incur any call charges.
The format of yourthe AAISP number should be in +44 format, e.g. to call AAISP support you'd use:
*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 of the number (which may confuse some caller ID display units), and not passed on if the call is diverted.
*So, for example, without any configuration or account details a softphone may call your A&AAAISP 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==
Most SIP clients which support calling SIP URIs are softphones. When testing, some clients can call SIP URIs without any configuration, other clients can call SIP URIs but require that a SIP account is defined (and perhaps enabled) although they don't actually use the account to place the call. There are also other clients which ''appear'' to support calling SIP URIs but which strip off the domain and call the number part as a chargeable phone call.
For testing clients, it can be useful to have a free SIP account which can't place external calls (e.g. [https://www.linphone.org/freesip/home linphone]) - that way one can be sure that the client isn't making chargeable calls.
==Known working clients==
You may need to tell some of your contacts how to call you using a sip: URI. To help, the following SIP clients are known to work:
* Ekiga (Linux)
* Linphone 4.0.1 (Android)
* Twinkle (Linux) - although currently (Dec 2018) there's a problem with SRV records when using systemd-resolved (on Ubuntu and Debian at least). Need to do
systemctl stop systemd-resolved
systemctl disable systemd-resolved
systemctl mask systemd-resolved
add the line
to the ''main'' section of ''/etc/NetworkManager/NetworkManager.conf''
and then
rm /etc/resolv.conf
to get NetworkManager to take over control of resolving.
* Voipfone Softphone (Android) deserves a mention. It needs an account with voipfone.co.uk, but you can signup and try for free. Their softphone app works out of the box for calling SIP URIs - provided they don't have numeric usernames: so you can call AAISP's office (e.g. support@aa.net.uk) but you can't call sip:<number>@host (it'll call the number direct instead).
==Known not-working clients==
* Linphone 3.6.1 (Linux) - doesn't use SRV DNS records, so can't contact @aa.org.uk SIP URIs.
* CSipSimple, GS Wave - need a SIP account to be enabled, which they then use to place the call.
* Zoiper (Android) - appears at first to work, but closer examination shows that it uses the SIP account to place calls (even when that account is disabled)


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