To add extension features, select an Extension and click the “Edit“-button. There are nine tabs on the top of the screen.
- Extension Routing
- Follow me
- Geo CID
- Advanced Settings
Extension features are described in the next paragraphs:
The Calendar feature allows users to link to an external calendar to setup a calendar-based calling behavior. That calendar is synchronized.
Supported calendars are:
- Microsoft Exchange Web Service Calendar
- Microsoft Exchange calendar
- CalDAV calendar
- iCal calendar
Destination if appointment
The “destination at appointment” determines the forwarding rule for incoming and internal calls. If there is an appointment entry in the calendar, calls are routed to the destination. If no forwarding is set, an absence tone is given. The Calendar distinct between sub-status 0=Free, 1=Tentative, 2=Busy.
|Name||Unique name of the defined calendar|
|Type||Choose the type of calendar|
|Calendar URL||Enter the URL where to find the calendar to check for events.
Support for HTTP redirect is currently not implemented. It is therefore important to enter the full URL and pay attention to the required slashes.
|Username||Username for accessing the calendar.|
|Password||Password for accessing the calendar.|
|Refresh time||The period in minutes between each calendar data synchronization.|
|Timeframe||The time span in minutes of the calendar that is covered by the synchronization.|
|Enable notification||Selects if a notification call to the user should be executed.|
|Minutes before event notification
|The amount of time, in minutes, that Ayrix will notify the user before a calendar event occurs. The user is notified of the appointment by an automated call with a voice message. If set to 0, the notification is disabled.|
|Destination if appointment|
|Destination||Select from the list the possible destinations.|
Housekeeping: These states are compliant with Estos Calendar replicator. Other substates are not (yet) supported. At SNOM we have a ”Presence” button, to indicate presence status: free or busy but it has no function change.
Voicemail is used for recording an audio message when the called party is not available. To do so, the user Interface Voicemail offers to configure the voicemail feature, as well to manage the recorded messages.
Voice mail messages can be listened by phone or sent by e-mail as a .wav file attachment.
A second way to manage voicemail features is accessing your voicemail box by dialing your own number.
A third method accessing the voicemail box remotely by dialing the destination, wait for the greeting message and dial “*” followed by the pin code. See next subchapter. Voicemail terminal menu guidance
The Voicemail announcement language is by default English. Set your preferred language in tab “Advanced Settings” – “Language”.
The window contains a configuration panel on the left and the Voicemail administration panel on the right.
|Enable Voicemail||Activate or deactivate the Voicemail function for the extension.|
|Timeout||The time limit in seconds before the incoming call is forwarded to voicemail box, if there is no answer AND no other forwarding’s are active. For ex. extension routing.|
|PIN||A PIN code number to save the voicemail settings The PIN is mandatory to save the record.|
|Voicemail if busy||Activate or deactivate call forwarding on busy extension
|Manage Greetings||Remark: To activate a next greeting message, delete the first message, then select the new Message and press save to activate.|
|Manage Greetings||Individual user-based announcement recordings|
|Announcement when busy||Activate or deactivate voicemailbox on busy extension|
|Announcement while not available||Announcement when your phone is not answered, disconnected or de-registered
|Temporary announcement||Temporary announcements. A temporary announcement has the highest priority. If set, it overrules all regular announcements|
|Name Announcement||To create a voicemail welcome greeting with your name|
|Number of messages||Shows the maximum number of new voicemails that will be saved in the list of new voicemails.|
|Delete after send:||Choose whether message recordings should be deleted after sending or downloading.|
|Delete oldest messages||Erase the last message from the new voicemail list.|
|Send attachment||Send voicemail messages as e-mail attachment (*WAV).|
|Play messages||Enable to play messages of the new and old voicemail list|
|Read out number||Speaks the message number prior to the voicemail message|
|Announce date and time:||Speaks the date and time before the voicemail message|
|Options for caller|
|Confirm voicemail||The caller can confirm if his recording is correct or wants to be re-recorded.|
The first two folders can be managed on Ayrix in the administration window.
The list of incoming voicemails, able to play, send or delete. When the voicemail is played or send, the voicemail is moved to the old message list, or being deleted when option “Delete after send” is set.
|Old voicemails||The list of archived voicemails, able to play, send or delete|
Voicemail list Panel
Voicemail terminal menu guidance #
Dial your own extension number on your phone to listen to your voicemails.
You have two options how to listen your voicemails. The first option is that you dial the number. The voicemail replies about stored voicemails, after you are guided through a navigation menu to handle them. The second option is that you dial remotely to the destination, when the voicemail starts you interrupt by pressing “*” and enter the Pin (password).
The terminal menu guidance offers you to record and activate new messages on the phone. These recorded messages are not! visible on the Ayrix “Voicemail”.web page..
Fig. Call flow Voicemail
Extension Routing #
The extension routing module allows you to forward calls in specific states of the extension. The destination can be an alternative destination, a voice mail, an announcement, or just busy tone.
Ayrix distinguishes between internal call forwarding and external call forwarding. In this window you configure call forwarding for those specific statuses.
|Inbound: Internal||Forward to destinations for internal calls|
|No Answer||Forwarding on no answer due ring time-out (in seconds) to the prefix number.|
|Busy||Immediate forwarding to prefix number when you are busy on the telephone and the maximum channels has been reached (the caller will be queued)|
|Not available||Immediate forwarding when the telephone is switched absent, in do-not-disturb mode, not registered, or not connected.|
|Timeout||The time-out in seconds before the incoming call is forwarded to the defined destination.|
|Prefix||The prefix number that points to the destination.|
|Inbound: External||Forwarding to destination for inbound calls|
|No Answer||Forwarding of an inbound call on no answer due ring time-out (in seconds) to the prefix number.|
|Busy||Immediate forwarding of the inbound call to the prefix number when you are busy on the phone, and the maximum channel occupation has been reached (the caller will be queued)|
|Not available||Immediate forwarding when the telephone is switched absent, in do-not-disturb mode not registered or not connected.|
|Time-out||The time-out in seconds before the inbound call is forwarded to the defined destination.|
|Prefix||The prefix number that points to the destination.|
|Outbounds||The list of outbound routing tables|
|+New||Add a new routing entry|
|Edit||Edit an entry in the config panel|
|Duplicate||Duplicate an entry as a basis for a new outgoing rule|
|Pattern||Enter the part of the trunk access number that differentiates between routes|
|Prefix||The prefix number that points to the destination.|
|Strip||Strip a (sequence of) digit(s) from the route pattern|
|CID||Depending on the route, display a CID, e.g. For example, with a “company internal dialing” you only want to show the caller the internal extension number instead of the public caller ID.|
Not only for inbound, but also for outbound routing we can configure extension-based routing rules. For example: A “priority or emergency trunk. only available for the CEO or chief officer. Whenever the first trunk is occupied or blocked, he is can dial out over a 2nd priority trunk.
“Outbounds” extension routing rules have priority over the general Outbound routing rules, (See chapter Outbound).
This Outbound panel is to configure an outbound routing table per Extension. In that table you can reconfigure a dialing digit numbers that are sent over a trunk by adding or stripping digits from the dialed pattern.
For example, anyone dials a number including country code in the own country and your provider only accepts national numbers in the country. A digit string of “00-41-56-520-xxxx” is sent. On the pattern “4156” we strip the country code for Switzerland “41”. Because of the country settings for national numbers a leading zero is placed. So finally, 056xxxxx is sent., see System Language & Region. Or in case you want to select on a specific trunk, for Least Cost Routing (LCR) or “In-company” dialing. Creating outbound rules, based on the dialed digits, you can select a particular trunk due .
For example, on 00-4156-520-xxx (Switzerland) you select Trunk 1, on 00-31-26-311-xxxx you select trunk 2 (Netherlands). We need to define a trunk access code to distinct between the trunks.
For example, selecting trunk 0 will we need a digit “8”, selecting Trunk 2 we need a digit “9” We add a prefix in front or our Pattern. “4156” we add a prefix “8” and for “3126” we add a prefix “9”.
With the Follow-me function, the user can create a ring group for his own extension (i.e., fixed and mobile numbers). A forwarded call will ring on all these destinations. The forwarding is active on all incoming channels.
The difference between a ring group and Follow me is, when a call is being answered on an internal destination, the originated extension is also occupied. In a ring group the originated number is free. An exception is, when the incoming call is answered by a mobile extension.
The user can add extensions to the Follow me group by clicking the “New”-button.
You can add external destinations, by selecting from the external number list, and enter the number in the right lower corner of the selecting panel. Enter your external number (including trunk access codes) and press “OK”
|Destination||The list panel of destinations (multiple possible)|
|New||Opens the selector list of extension or external number|
Housekeeping: By a follow me call answered on the destination, the caller list on the originator telephone is updated with SIP response “Answered elsewhere”.
Geo CID #
Geo CID allows an Extension user to send a country-specific Caller ID, depending on the E164 country and region settings. The Ayrix is limited to one (“1”) Geo CID table. This Geo CID allowance list can be configured in the menu PBX > Geo CID. This feature is useful where your company has branch offices in different countries or regions and want to serve your customers from headquarters. Per extension, this list can be either enabled or disabled.
|ACTIVATE_GEO_CID||Selects if a Country/Region dependent Caller ID is sent to the opposite party.|
|Enable Geo CID||Enables / disables Geo -CID caller indication.|
|Configure CIDs||The list to configure country/Region specific Geo CIDs. The Geo-CID rules are shown. Configured in the PBX Geo CID rule configuration panel.
Now add the specific CID.
Important Notice: Be aware, Ayrix has different! Geographical based functions:
- Geo CID: Send a country/region specific Caller ID (CID)
- Geo Restrictions: Restrict incoming and/or outgoing calls with specific Caller- ID/Calling ID.
Advanced Settings #
This feature allows the user to enter additional settings for his extension.
|NAT||NAT: Default -enables Ayrix STUN -stunning server. Depending on the IP address the SIP messages are passed directly to the Ayrix or it is passed to the STUN server. *see chapter NAT: explanation
No: switch off the STUN server. The SIP header is not modified, used in case of Ayrix and Extensions are in the same IP subnet. For. Ex: Ayrix ISO is directly installed on hardware.
Force: forces the full SIP signaling traffic passing the STUN server.
|Directmedia||Directmedia enforce media streaming via P2P
No: RTP media stream passes the PBX
Yes: Enforce RTP media stream P2P
No NAT: send RTP media stream P2P only in the same subnet
Update: send an update Element to switch over from PBX to P2P
|Music on Hold||Selects type of MoH, selecting a Sound File or Radio Stream.|
|DTMF Mode*||See chapter DTMF Mode|
|Language||Select the Language for voicemail and country specific media, ringtones e.g. based on the individual extension.|
|Type||Specify the call direction.
The following table shows the respective functions:
Ayrix <= user
Only outgoing calls possible.
Ayrix => peer
Only incoming calls possible.
Ayrix <=> Friend, both way traffic.
A peer can only be called, a user can only call, and a friend can do both.
The opposite party gets a disconnected, E503 Service unavailable (-tone), or when
voicemail is enabled on the extension, the calling/called party receives an “Unavailable” message, and the caller can leave a message. (normal call completion).
|Recording||[Yes/No] Record your phone calls when set to “Yes”.|
|Member of pickup group(s)||Add the extension to a pickup group, getting member of it. You can pick up calls from other members by pressing [*8] and going off hook. Multiple group membership is possible.|
|ACCEPTOR_GROUP||A Member of an acceptor group can pick up calls from your extension, even you are not member in that specific acceptor group.|
|Video Support||Select if your extension supports video calls|
|Text support||Select if your extension supports text messaging on the phone display.|
|Header 1||A 1st text header template to send additional text messages.|
|Header 2||A 2nd text header template to send additional text messages.|
|Maximum incoming lines||Limit the maximum incoming lines (1…99) on your extension number. If you select more as one (1) line when a 2nd incoming call, a call waiting tone is sent to the called party. If the maximum lines are occupied the next caller gets busy tone.
Default is set to “1” line.
|Number of simultaneous channels||Limits the maximum channels (1…99), used for incoming and outgoing calls simultaneously. For example: series calls. Make calls while a 1st, a 2nd, 3rd called party is parked. When the maximum of channels exceeds, a display message is shown <Temporary unavailable>, E480.
<empty> = number of channels as needed (max. 99)
|Deny registrations from IP / Network||Blacklist IP address or IP range to prevent telephone registrations on this extension nr.|
|Allow registrations from IP / Network||Whitelist IP address or IP range to allow only telephone registrations with that IP -range this extension nr.|
|Deny calls from IP /Network||Blacklist to prevent phone calls from this IP address or IP range on this extension nr.|
|Allow Calls from IP / Network||Whitelist to allow only phone calls from this IP or IP range on this extension.nr.|
If the caller is in a private subnet behind a NAT gateway, the “Invite” request contains multiple private IP addresses (Listing 2) that cannot be reached by the other side.
The gateway, which carries out the address translation on layer 3 (IP) or on layers 3 and 4 (TCP / UDP, NAPT: Network Address Port Translation) in the packets, is inherently unable to handle IP addresses in higher layers to recognize and modify, but a STUN, -stunning is server is capable to translate these SIP IP addresses from private into public.
The Ayrix can be installed as virtual machine behind a router and firewall, the Ayrix does have a “build-in” STUN server and NAT traversal for this operation. See also System – Network – tab Connectivity.
SIP offers multiple methods to transmit Dual Tone Multi Frequency (DTMF) tones to the opposite party during the call. The DTMF Mode selects the protocol of digit- transmission during a call. These DTMF tones can be detected by the IVR or a contact center at the opposite site.
Digits can be transmitted as in-band (tones) in the media stream, that you will hear, or transmitted as Info- or short info element, in the SIP message, (outside the speech channel, you will not hear anything).
|RFC 2833: in band DTMF tones
Info, DTMF send outbound as SIP message protocol
Short Info, DTMF send as outbound short-info element in the SIP protocol
Auto: depending on the offered services at the called party site, the system selects the appropriate mode
*Notice: being interoperable sending DTMF, both sites, the Ayrix as well the SIP terminal needs to be configured correctly. Whenever your choice of DTMF tape, In-In-band, RFC2833 or SIP_INFO element, both sites, Phone and PBX, need to accept that mode. Example: Basic settings of DTMF on a SNOM D315: Identity “Number” –> SIP –> DTMF via SIP INFO –> on –> Apply
Example: Basic settings of DTMF on a Grandstream G2515:
See also Section: SIP terminal features
To get a transmission of your voice, the telephone needs to convert your voice into a Media stream or RTP.
Due call setup the SIP protocol negotiates about the codec to be used between the parties. The first codec that matches on both sides is chosen. Through the years several codec flavors have been developed. If you assigned a SIP trunk on with a Codec selection, already on beforehand the extensions got already that default codec selection available. However, for every Extension you can create a preferred Codec selection. The codec window shows a “Available Codecs” list and a “Selected Codecs” list. You can move and rank your selection from the available to the selected list.
|Selected Codes||The selected and ranked list of codecs for Extension|
|Available Codecs||The list of available Codecs|