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Automatic IP phone detection
AskoziaPBX supports automatic detection of supported VoIP phones in your local network (LAN) as an alternative to Auto-Configuration. This is very comfortable as you don't have to manually enter your phone's MAC address.
|Automatic phone detection works only for phones officially supported by Askozia. A full list is provided in the Supported Hardware chapter|
|The IP phones to be automatically detected by AskoziaPBX, need to support SIP multicast. Additionally, reset your phone to factory defaults before starting the phone detection. If a supported phone is still not detected, make sure to use the firmware version as stated in the model list of the auto-configuration settings and in the Supported Hardware chapter.|
The following video provides a tutorial about how to use automatic IP phone detection.
In case your phone isn't recognised automatically, you can add it by using Auto-configuration and entering its MAC address manually.
The general settings allow the basic configuration of a phone.
- In the number text box, the extension of the telephone is specified.
- The caller ID is the name or number that dialog partners see on their phone displays.
- Language sets the spoken language of the audio prompts. This setting defines which prompts are used internally. The language setting on a provider account determines the language being heard when external calls come in and overrides this settings.
- A description for the phone can be added. The description is ignored by AskoziaPBX but may be handy to clarify a phone's purpose, user or location.
- In order for the phone to register with AskoziaPBX, a secure password is generated automatically. You can either use this password or choose a different one. This password needs to be used for both telephone and AskoziaPBX phone account as the phones otherwise can not register and will not be able to place or receive calls.
- For public access, check allow this number to be reached over the Internet. You can use a friendlier alias instead of the extension number. A possible form could be myName@myIP.
- Block providers allows to prevent access to the selected provider. This may be useful if a telephone should not have access to a certain provider to save costs. If all providers are blocked, the telephone can only be used internally.
- Secure calling can be activated for this phone after it has been configured in the Secure Calling section of the Connectivity menu.
For officially supported IP phones, the Auto-Configuration settings are automatically transmitted to the phone after saving your configuration.
First, the provisioning template for a supported phone model needs to be selected in order to cover the basic configuration of the phone.
If an unsupported IP phone is to be used, the selection is to be left empty and the configuration to be done manually in the web interface of the phone.
By entering its MAC address, the phone will automatically be configured with the same settings as this phone account, avoiding additional manual configuration in the web interface of the phone itself.
The MAC address of your phone is usually located on its bottom side or in its web interface.
|To gain access to the web interface of a snom phone, first push the help button of the phone.|
In the Speed Dial section, numbers can be assigned to the speed dial buttons and busy lamp fields (BLF) of the phone if existing. Please make sure that you only assign as many numbers as your phone has speed dial buttons.
Click on Save to save your configuration. Then, restart your phone by disconnecting it from the power supply for a moment. After the restart, your phone is configured and registered with AskoziaPBX.
By pressing the alt-key, the following additional options appear. However, in most cases, they are not required.
- Show template
- Show phonebook
- Show dial URL
AskoziaPBX supports auto-provisioning for a variety of IP phones. Details about the supported models are provided in the Supported Hardware chapter. Below, you can find some examples for auto-configuration.
How to connect external IP phones via VPN
We strongly advise to not assign a public IP to your AskoziaPBX phone system as it can cause your PBX to get hijacked and may result in high phone bills! Please always use a firewall both for your PBX and your network, and connect external IP devices via VPN tunnel!
To automatically configure IP phones via VPN, you either need a layer 2 VPN (as explained in Auto-configuration) or a layer 3 VPN. For layer 3 VPN, follow the instructions below. In this example, we are using a SNOM 320.
1. Open the SNOM web interface and got to Setup -> Advanced -> Update
2. Open a second tab and go to Askozias web interface Accounts -> phones -> (your desired phone) -> Auto-configuration and press the alt-key -> two links will appear
3. Click on show template with the right mouse button and copy the linked URL
4. Switch to the SNOM web interface and enter the copied URL into the field setting URL
5. Further activate the Update Policy always update
6. Apply your settings and Reboot
|Do not click on reset here|
The phone will now reboot and download the XML file.
The same steps have to be used for provisioning Yealink phones via VPN. The web interface of a Yealink T46G is shown in the picture below.
The video below provides further information about how to set up VPN-tunnel between your PBX and an external phone.
How to configure a soft phone
The following video provides an example of how a soft phone can be registered and configured with AskoziaPBX.
Call Control CTI
Call Control CTI allows you to control your IP phone with your PC, Mac or mobile device via a web browser.
If a CTI license is activated, every phone account can make use of it’s features by default. CTI features can be deactivated by ticking Disable CTI features for this account.
In order to get a certain user started with the CTI, login information can be sent via e-mail by clicking on Send user instructions within the respective phone account.
Further information about the CTI are provided in the Getting Started with the CTI chapter.
Client User Interface
Activate Client User Interface allows to give limited access rights to the user of a certain phone account in order to make modification in Askozia’s web interface.
In order for users to log in, both a username and a password need to be assigned.
Under Allowed User Actions, specific access rights can be assigned to the user of the respective phone account.
- Check/Uncheck all: activates or deactivates all of the possible access options listed below
- change the caller ID in the General Settings
- change the language for CTI and audio prompts in the General Settings
- Edit speed dials for this account in the Auto-configuration
- Change call notification settings
- Change voicemail settings
- Change settings for parallel call
- Change settings for secretary feature
- Print dialplan in Dialplan
- Access fax archive in Services
- Delete fax archive in Services
- Access call detail records in Status
In User Notes, the system administrator can leave notes that will be displayed in the interface of the client user after activating show the notes below to the user.
After activating the client user interface for at least one phone account, the User button appears on the landing page of Askozia’s web interface. After entering the login data as specified above, the user can access the web interface and modify his settings according to his access rights.
Beside the options specified above, users can be granted access rights to Stateboards, which can be accessed through the client user interface as well.
Call notifications and voicemail
- Ring length defines the period in seconds before hanging up or going to voicemail. Indefinitely disables any interference by AskoziaPBX. In case notifications are not configured (see Notifications) but a ring length has been defined, AskoziaPBX refuses the call by signaling "unavailable" after the end of the time.
- In case of the specified ring length being met by an incoming call, forward on timeout specifies respectively for internal and external calls if incoming calls should be forwarded to voicemail, another internal extension or call flow.
- In case of the callee being already busy with another ongoing call, forward on busy specifies respectively for internal and external calls if incoming calls should be forwarded to voicemail, another internal extension or call flow.
- In case of the callee being unavailable or unregistered, forward on unavailable specifies respectively for internal and external calls if incoming calls should be forwarded to voicemail, another internal extension or call flow.
- If you want to receive notification e-mails for missed calls, select notify me of missed calls via e-mail. Enter the e-mail address where notifications should be sent to in the text box below.
|To receive notifications, call recordings or voicemail via e-mail, the e-mail server settings need to be correct. To confirm this, click E-Mail Notifications. More information on this in the Notifications section of the Services chapter.|
- To enable voicemail, select enable voicemail to e-mail and enter an e-mail address. Once enabled, all incoming voicemail messages will be sent to your e-mail account as audio files.
- To also receive voicemails on your phone, select save voicemail to storage to access it from phone and enter a PIN. Dial 000086 and enter your PIN to listen to your voicemails or record your personal voicemail greetings.
|Personal voicemail greetings are only played back to the caller if no global voicemail greeting is defined in the System#General_Setup of the System menu.|
|To listen to voicemail on your phone or receive call recordings, additional storage (e.g. a flash drive) needs to be connected to the telephone system.
More on that in Storage.
Voicemail only works if a maximal ring length is specified. Using indefinitely automatically disables voicemail.
- You can upload your own unavailable and busy messages by browsing and upload the desired audio files. Activating Remove file deletes the corresponding file after saving your configurations.
Outgoing caller ID
The outgoing caller ID for each phone can be set for each provider. Leave the field blank to use the default settings of the respective provider. Use the formatting "Your Name <number>". For some providers, the country code (such as 49 for Germany) and/or area code (such as 30 for Berlin) may have to be used as prefix
|In order to correctly set the outgoing caller ID, the option Clip-no-Screening needs to be booked from the respective provider.|
|Depending on the provider, country and area code may have to be put in front as prefix.|
Parallel Call and Secretary features
The extensions named in Parallel Call will ring simultaneously. Enter the desired numbers separated by white space or leave this field empty. This feature is limited to 10 phones.
If call forwarding is activated on your phone, the phones named in the Secretary field will be excluded and their calls still be routed to your phone. This feature is especially useful for communication between manager and secretary. Enter the desired numbers separated by white space or leave this field empty. This feature is limited to 5 phones.
For the example shown in the image below, your phone would ring simultaneously with the extensions 102, 103 and 110. Furthermore, calls from the extensions 102 and 103 would be routed to your phone independently of call forwarding being activated or not.
AskoziaPBX supports a variety of free and licensed Codecs for both audio and video. Codecs are explained in detail in Help for Integrators for experienced users.
AskoziaPBX offers two ways to create call recordings. Manual call recordings can be initiated by pressing *1 during a call. Automatic call recording records all calls without user intervention.
|To receive call recordings via e-mail, the e-mail server settings need to be correct. More information on this in the Notifications section of the Services chapter.|
Manual call recording
Activate send me recordings via email enter a valid email address to which the recordings are send to. To record calls, simply press *1 on any other VoIP (or the record button on your Snom phone). Recordings are sent to you as a .wav file after pushing *1 again or hanging up.
Manual Call Recording should be used for maximal 10 minutes recording time. Since the recordings are send via email, the amount of data will be too large with longer recordings.
The following video tutorial shows how to use manual call recording.
Automatic call recording
Automatic call recording can be activated globally for all phones or for individual phones in their respective accounts. First, make sure to enable global call recording in the call recording settings of the Miscellaneous section in the Advanced menu.
Afterwards, call recording can be activated for individual phone accounts as shown below.
The video below introduces automatic call recording and archiving.
Hot desking is used in companies with shared office infrastructure. In case of desks being shared by multiple employees, it has to be possible that the users can switch their account setup (voicemail etc.) from one phone to another. Hot desking makes this possible.
Please note that hot desking only works for phones, auto-provisioned by AskoziaPBX
To activate hot desking in AskoziaPBX, the check box enable hot desking for this account needs to be activated and a password for this account needs to be set.
After configuring the hot desking for this account, the following code has to be entered on the desired phone.
*0*XXX (extension of the phone account to be used for Hot Desking)
Now, it's required to enter the previously specified password and confirm it by pressing the pound key. The telephone will reboot and load the desired account. If the desired account has been active on another phone before, the accounts on both phones will switch.
By entering *0 on a phone, an accounts can be deactivated again.
Advanced options are additional parameters not required in most cases. The following parameters are available.
- When entering a caller ID in the overwrite caller ID text field, this caller ID will be displayed on display of other phones. This can be useful if you are using multiple phone for the same user. Use the format "name <number>"
- A manual context name can be defined for this phone account and be used within SIP configurations and dial strings.
- Send Remote-Party-ID sends the Remote-Party-ID in the packet header and provides information about the remote party. Choose the required mode from the drop-down menu.
- If activated, Trust Remote-Party-ID will copy the number from the RPI header instead of the "From" header of the request it received on the inbound call leg.
- To enable and set the ISDN transfer capability of a call, choose the required mode from the drop-down menu.
- 0x00, speech (default, voice calls)
- 0x08, unrestricted digital information (data calls)
- 0x09, restricted digital information
- 0x10, 3.1kHz Audio (fax calls)
- 0x11, unrestricted digital information with tones/announcements
- 0x18, video
- Activating AOC passthrough enables passthrough of AOC (Advice of Charge) messages to send fee information to this phone.
- Allow auto-answer may be useful if the phone account is associated to an intercom.
- Night Switch exception makes the phone available from the outside, even when the night switch is activated.
- Phone Book exception makes this phone unavailable in the local phone book that is provided to all auto-configured phones registered with Askozia.
- Default provider determines one provider this phone uses for external calls. Please note that this overwrites your outgoing dial patterns.
- By activating SIP Permit/Deny, the SIP traffic to a certain IP range can be limited.
- For example, setting SIP Deny Network to "0.0.0.0 / 0.0.0.0" denies all SIP traffic.
- For example, setting SIP Permit Network to "0.0.0.0 / 0.0.0.0" permits all SIP traffic.
- T.38 Gateway capability should be activated if one side of a call is not capable of T.38 mode. T.38 is used to transfer faxes over data networks in real-time. In order to do so, T.38 converts the analog fax signal into an image and transfers it to a T.38 compatible device over the network. If this device is an IP phone system or a gateway, it can transform the signal back from T.38 into an analog signal and send it via PSTN (the public telephone network) to an analog fax machine. If both communicating sides are capable of T.38 mode but the gateway mode is still activated, the gateway won't interfere and allow transparent T.38 pass-through.
- NAT offers several NAT modes. Choose the required mode from the drop-down menu.
- always use NAT mode
- only use NAT mode according to RFC3581 (rport)
- never attempt NAT mode or RFC3581 support
- assume NAT mode but don’t send rport
- DTMF Mode specifies how DTMF signals are transferred via SIP. Choose the required mode from the drop-down menu.
- Qualify frequency defines how often the availability of the peer will be checked (by default every 60 seconds).
- Qualify defines the time period in seconds after a packet has been sent, before the peer is considered being unreachable. The default value is two seconds.
- At ringtone, you can specify if you prefer a local ringtone instead of the alert-URL.
- Busy level defines how many parallel calls the phone can handle before signaling "busy".
Manual dialplan incoming/outgoing
Manual dialplans can be used in different contexts but are only valid for the context that they are specified for. Therefore, the dialplan defined for an external phone is only valid within the context of this particular phone.
Blocking outgoing numbers
To block numbers, such as special service numbers, the following commands have to be entered in the Manual dialplan outgoing field of the phone. The following picture shows blocked 0800 and 0900 numbers and those, starting with 01. The commands can be modified at the marked positions for different numbers.
Configuring 00.+, blocks all international prefixes.
Exception for blocked numbers
Blocked numbers in a provider account cannot be called from all internal phones via this provider. If exceptions for single phones are required, the following command has to be entered in the Manual dialplan outgoing of the desired phone account.
Manual attributes are only required in a few special cases and should generally be used with caution and only by experienced administrators.
As manual attributes are only valid in specific contexts, manual attributes for an SIP phone are only valid in the context of this particular phone.
You can find an overview of all existing manual attributes in the documentation of Asterisk.
Save and reboot
In order for all settings and changes to take action, you have to click on save in the end. By default, the IP phone assigned to this account will be rebooted afterwards. It is possible to suppress the phone reboot. However, it is recommended to allow the phone to reboot in order to activate all settings.