autoreview, Bureaucrats, editor, emailconfirmed, Administrators
8
edits
Incoming VoIP Features: Difference between revisions
m
Removing reverence to Special, "unsupported" tag feature:
m (clean up, typos fixed: along side → alongside, eachother → each other, ie → i.e. (3), , → , , etc → etc. , eg → e.g. (8)) |
m (Removing reverence to Special, "unsupported" tag feature:) |
||
| (34 intermediate revisions by 4 users not shown) | |||
|
[[Category:VoIP]][[Category:Mobile]]▼
[[Category:Control Pages]] ▼
▲=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.
===Voicemail===
===SIP Phone===
*You can register multiple sip phones to our server
===Tag===
The Tag is 4 characters that will be prefixed to the 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. E.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']]
[[File:Snom821-tag-screenshot.png|none|frame|The tag is set to 'aTag']]
The above screenshots are from a Snom 715 and a Snom 821 showing incoming calls with a tag of "aTag".
The SIP for a tag is shown in the From field:
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===
An example of using srv records would be as follows: Say you have two VoIP servers and they have the public IPs of <code>192.0.2.50</code> and <code>192.0.2.60</code> and you want to give it the DNS name of <code inline>a-pbx.example.com</code> and <code>b-pbx.example.com</code>, and then use <code>pbx.example.com</code> as the SRV record, you'd create the following DNS records for it as follows:
a-pbx.example.com. A 192.0.2.50
b-pbx.example.com. A 192.0.2.60
_sip._udp.pbx.example.com. SRV 1 0 5060 a-pbx.example.com.
_sip._udp.pbx.example.com. SRV 2 0 5060 b-pbx.example.com.
In the AAISP control pages, you'd enter <code>pbx.example.com</code> as the server hostname, our systems will then look up the SRV records and will route the call accordingly.
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', 'host' or 'nslookup' on the command line, e.g.:
$ dig +short srv _sip._udp.pbx.example.com
$ host -t SRV _sip._udp.pbx.example.com
$ nslookup -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===
Simple Call Gates are supported, where by a number can be called and a message played back such as 'Please press 1 for Sales, 2 for support...'. The system will then put the call to a corresponding number.
For more information see: [[VoIP - Call Gate]]
=Outgoing Tab Features Overview=
===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===
==Other Control Page pages==
<ncl style=bullet maxdepth=5 headings=bullet headstart=2 showcats=1 showarts=1>Category:Control Pages</ncl>
| |||