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=Related Pages on the A&A Website:=▼ [[Category:VoIP]][[Category:Mobile]]▼ [[Category:Control Pages]] ▼ ~~[[Category:VoIP Features]]~~ ▲=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] =Trunk and PBX Features= Our VoIP system can simple 'trunk' calls to/from your own PBX - it is perfectly normal for you to have your own office PBX, such as a FireBrick or an Asterisk server for example and for your equipment to provide you will all the 'PBX' type functionally that you require. However, you can also have SIP phones register directly against our servers and use the PBX functionality that we provide. Or, indeed, have a mix of the 2 - ege.g. we can route calls to you by SIP (trunk), but we can also record calls. ---- =Incoming Features Tab Overview:= These are the features on the Control Page, under the 'Incoming' Tab. Recordings are in stereo - with the 2 parties on separate channels. Callers can be warned, with a pre-recorded announcement, that the call will be recorded [[~~VoIP_~~VoIP -~~_Recording_Calls~~ Recording Calls\|Read More]] ===Queuing=== ===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=== a number can have multiple time ===Multiple Targets=== Incoming call routing is configured on the Control Pages. Call Routing is based on setting the 'Target' - a number can have multiple targets, ege.g., can be routed to a SIP phone as well as a mobile, and multiple other numbers. Indeed, you can also register multiple SIP phones against the same number, and calls will go to all the SIP phones that are registered. ===Delays=== Each target can be given a delay (~~0-30~~0–30 seconds). EgE.g., you can have your number ring your SIP phone immediately, and then also ring your mobile after 10 seconds. ===Number Announce=== Put a number in to here, and a message will be given to callers that the number has changed and the number entered will be read out as the new number. ===Fail=== Number to call if the call fails to get though to the configured endpoints (iei.e. a registered phone). ~~Fail number doesn't get used~~i.e., if ~~you~~the ~~have~~registered ~~voicemail~~phones ~~set~~are ~~for a number~~unavailable, ~~as the~~then call ~~will~~this gomobile ~~to voicemail~~number. This can be used alongside voicemail, in this case, the call will go to voicemail and have the message played but rather than having the option to record a message the call will be transferred to the fail number. Ie: Incoming call --> (optionally ring phones) --> Play Message --> Divert to Fail number This is useful where you want to play a message before ringing someone else - e.g. 'Thank you for calling, we're redirecting you to our on call engineer now. ===Transferring Calls=== Transferring calls is supported, and with VoIP this is usually handled by your telephone. iei.e., your phone would have a Hold or Transfer button, which enables you to either Blind transfer or perform an assisted transfer. ==Targets in Detail== ===SIP Phone=== You can register multiple sip phones to our server., ~~(single~~and ~~phones~~calls ifwill ~~you're~~be onsent ~~the~~to ~~legacy~~all Aregistered ~~server)~~sip phones. The Tag is 4 characters that will be prefixed to the number and shown on your SIP phone display (if it has one) ===~~Mobile~~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. ~~If you have a SIM with us, then the call will also be sent to the SIM~~ [[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. ===~~Also~~More ~~Ring~~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: These are up to 10 other numbers that we'll send the call to. They can be other numbers you have with us, or can be any other number. eg mobiles, international etc - any number which you can normally dial from your account. The charge for the call will be the same as if you were dialling the call normally from your account.▼ a-pbx.example.com. A 192.0.2.50 ===Call Gate - IVR===▼ 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. 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.▼ 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> ~~To set this up:~~ 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] Set the 'Main' number can to always to go voicemail, or to go to voicemail after an amount of time. Fill in some of the Also-rings to go to the DDI of the various targets you want Enter a 'fail' number, as this will be used as the default number to call if no digits are keyed by the caller *Register a phone to the 'Main' number, dial 1571 and record your 'Please press 1 for Sales, 2 for support...' message You can test your SRV record using 'dig', 'host' or 'nslookup' on the command line, e.g.: This then works as follows, once the caller has called in and has been answered with the voicemail message, the user can key a digit. If there is an also ring number set up for the corresponding digit then the call is put through to that number as a divert. $ 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. If there is a fail number set up as well as some also ring numbers then the fail number is used as a default if no digits are keyed. If there is a fail and no also ring then the fail number is used immediately after the announcement. - this then works as a greeting before a call is put through to phones. ===Also Ring=== ▲These are up to 10 other numbers that we'll send the call to. They can be other numbers you have with us, or can be any other number. ege.g. mobiles, international etc. - any number which you can normally dial from your account. The charge for the call will be the same as if you were dialling the call normally from your account. ▲===Call Gate - IVR=== ▲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. ege.g. if you have 2 numbers <code>01344400001</code> and <code>01344400002</code>, then you can call ~~eachother~~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=== This is the outgoing Presentation Digits that are set when a call is made from this phone number. ege.g., we can set something other than their phone number here. There is a charge and a process for this and we'll need paperwork to prove the number is yours. Contact Sales for more details. ===Local Prefix=== This is an area code, (ege.g. 01344, 020) that will get prefixed to local numbers. Useful for people with an 03 number wanting to dial local calls without the area code ===IP Lockdown=== =VOIP Security= VoIP accounts can be compromised , so care is needed to this does not happen. Please see our [[VoIP Security]] page for more information. [[~~VoIP_Security~~VoIP Security]] ==Other Control Page pages:== <ncl style=bullet maxdepth=5 headings=bullet headstart=2 showcats=1 showarts=1>Category:Control Pages</ncl> ▲[[Category:~~Control~~Voice ~~Pages~~SIMs]] ▲[[Category:VoIP~~]][[Category:Mobile~~ Features]]