Incoming VoIP Features: Difference between revisions

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AA-Andrew

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Revision as of 17:14, 29 December 2018 (view source) Adsb (talk \| contribs) (Distinguish between caller's number and caller's name) ← Older edit		Latest revision as of 09:29, 15 March 2021 (view source) AA-Andrew (talk \| contribs) m (→‎Your Server)
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===ACR=== Anonymous Call Reject - callers who withheld their number will have a message played to them and call will be rejected. Call 114 to hear that message, and 113 to record your own message. An alternative way of handling Anonymous calls is to not reject them on the account's control panel and, if ''all'' configured targets reject Anonymous calls, and you have voicemail enabled, then Anonymous calls will go direct to voicemail. Anonymous Call Reject - callers who withheld their number will be rejected ===Voicemail=== ===SIP Phone=== You can register multiple sip phones to our server., and calls will be sent to all registered sip phones. ===Tag=== It's worthwhile checking that your local equipment can cope with your chosen tag. For example a phone with a seven segment display will have difficulty with some alphabetic characters, and some phones can only display 12 characters (and a typical UK phone number is 11 characters) so you'll lose some characters. ~~'''Special, "unsupported" tag feature:'''~~ If you forward a VoIP number to another number such as your mobile (using the Also Rings), the CLI presented to your mobile will be the caller's number (the caller's name field probably being lost). You won't know if this is a redirected call or not. There is a hidden feature where by we can add a prefix to the CLI. In the 'tag' box, set it to N= (e.g. 2=), then the CLI will be sent as +11N (in our example, +112) followed by the number calling. This way you will be able to tell that the incoming call is one being redirected from your VoIP number and you still get the CLI of the caller (the N= string is also prefixed to the caller's name field, but this is unlikely to be passed through the PSTN). ~~(This feature was added 2017-04-20, but may be removed or improved)~~ ===Your Server=== We will look up SRV records if there are any and will follow those. SRV records make routing SIP to you very flexible. SRV records are able to specify the SIP port used, in this case we will try that port. SRV records also allow you to specify multiple hosts with priorities. SRV records will help you create a resilient SIP system at your side by using multiple SIP servers etc. We only support UDP and not TCP at the moment, so your SRV records need to be for the UDP protocol. The accepted hostname formats are: hostname * hostname:port * ip4_addr * ip4_addr:port * [ip6_addr] * [ip6_addr]:port ===More about SRV Records===