Incoming VoIP Features: Difference between revisions

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=Related Pages on the A&A Website:= [http://www.aa.net.uk/telecoms.html www.aa.net.uk/telecoms.html] [http://www.aa.net.uk/kb-telecoms-sip.html www.aa.net.uk/kb-telecoms-sip.html] ---- =Incoming Features Tab Overview:= These are the features on the Control Page, under the 'Incoming' Tab. ===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=== The Tag is 4 characters that will be prefixed to the ~~number~~caller's name field and (perhaps) shown on your phone's display (depending on whether your phone's display shows caller's number, caller's name, or both), this can be used to help identify the number the caller called. EgE.g., the tag could be set to ''Sale'' or ''Tech'', and then you'll know what type of call you are receiving. [[File:Snom715-Tag-Screenshot.png\|none\|frame\|The tag is set to 'aTag']] From: "aTag07508xxxxxx" <sip:07508xxxxxx@voiceless.aa.net.uk>;tag=2015012110443100001 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. ===Your Server=== We can route calls to your own SIP server, fill in the details of your server here. We will try IPv6 and IPv4 if they are available. 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. ===More about SRV Records=== The format of the 'host' part of a SRV record is: <code> _service._protocol.name</code>. The format of the 'value' of an srv record would be in the format of: <code>priority weight port host</code> Like with MX records, lowest-numbered priority gets tried first, weight is used for records with the same priority. More info in RFC 2782 and on [https://en.wikipedia.org/wiki/SRV_record#Provisioning_for_high_service_availability Wikipedia] You can test your SRV record using 'dig' on the command line, eg:▼ ▲You can test your SRV record using 'dig', 'host' or 'nslookup' on the command line, ege.g.: $ dig +short srv _sip._udp.pbx.example.com 1$ 0host ~~5060 a~~-t SRV _sip._udp.pbx.example.com. 2$ 0nslookup ~~5060 b~~-type=srv _sip._udp.pbx.example.com. Not all computers have all these commands, Linux/Mac probably will, but on Windows try nslookup. ===Also Ring=== ===Centrex=== This allows you to use the last 1, 2 or 3 digits of your phone number to call other numbers in your block. e.g. if you have 2 numbers <code>01344400001</code> and <code>01344400002</code>, then you can call each other by using 001 and 002 if you set Centrex to 3. This is also used when you transfer calls between your numbers. ===Presentation===