Accounts

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AskoziaPBX organizes providers, phones, DECT bases and faxes in accounts. Every provider and every device has its own account to administrate the user settings. Accounts control access authorization, regional settings and technical configuration.


Contents

Providers

A provider account determines how calls through this provider are handled. Call routing for incoming and outgoing calls is defined and telephones are matched with their external call numbers.

Every technology has its own account type and specific settings. To add a new account or edit an existing one, click Providers in the menu bar. Choose the type (SIP, IAX, ISDN or analog) to enter the configuration menu.

Provider types

The provider overview lists all available providers. The Askozia bullet green.png in front of the name of a VoIP Provider indicates that AskoziaPBX is registered with this provider and you can make phone calls. Askozia enabled.png marks enabled providers, by clicking on it you can disable providers. Askozia disabled.png marks disabled providers, by clicking on it you can enable providers.

Provider overview


SIP Provider

General Settings for SIP Providers

In Pre-configuration, you may choose from a variety of providers in order to use the predefined settings provided by AskoziaPBX. If your provider is not included in that list, the configuration is to be done manually starting with the Name of the provider and the Host. The Host is either a URL or an IP address. Username and Password are required next to complete the basic provider setup.

From the Language list, the language for the audio prompts for this account can be chosen. The specified Public Number will be read back to the caller when reaching voicemail. By default, the account's username is used if it is numeric. If the username is not numeric, the internal extension the call was routed to will be read back. It is advantageous to enter your entire call number in this field to make sure the caller is not just told the extension.

General settings for SIP providers


Advanced Options for SIP Providers

Advanced Options are additional parameters, required by some providers. For most users, these settings are not required. The following advanced options are available for SIP providers.

  • Multiple Hosts: If this provider has multiple servers to forward calls to you, specify a host for each server.
Advanced SIP provider settings, part 1, multiple hosts
  • With Blacklists and Greylists you can prevent specific numbers from calling in. Blacklisted numbers receive the busy signal immediately, grey listed numbers are forwarded to the voicemail of the extension the caller tried to reach. Of course, this only works if voicemail for this phone was configured. Specify one black/greylisted number per line. Select Blacklist anonymous calls or Greylist anonymous calls if you want all calls with withheld number to be black/greylisted.
Advanced SIP provider settings, part 2, blacklist and greylist
  • Add PAI: Add P-Asserted-Identity header field as defined in RFC 3325.
  • Add PPI: Add P-Preferred-Identity header field as defined in RFC 3325.
  • Add RPI: Add remote-party-identity header field
Advanced SIP provider settings, part 3, header fields
  • Public Access: Allow this number to be reachable over the Internet. This option is required if the IP of your provider is changing with every call.
  • Register Separately: Register every incoming number separately. This is necessary when you don't have a SIP trunk and don't want to add a provider for every single number.
  • T.38 capability: If supported by the provider, T.38 mode should be used. 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.
Advanced SIP provider settings, part 4
  • SIP TO as DDI: Use SIP TO-Header as incoming DDI.
  • SIP CPI as DDI: Use P-Called-Party-Identity header as incoming DDI.
  • SIP Permit/Deny: activate permit/deny mask for SIP traffic.
    • SIP Deny Network: Limit SIP traffic to and from this peer to a certain IP or network. Set to '0.0.0.0 / 0.0.0.0' to deny everything.
    • SIP Permit Network: Limit SIP traffic to and from this peer to a certain IP or network. Set to '0.0.0.0 / 0.0.0.0' to permit everything.
  • Account code: This code is generated individually for each account. Usually there is no need to change it.
  • Registration: In some cases a registration is not permitted by a provider. Select Do not register with this provider if this applies to your provider. In most cases, AskoziaPBX is able to do the formatting of the registration correctly. Just a few providers expect a special formatting of the registration string. You can specify a registration string in the same format Asterisk uses in the text box.
Advanced SIP provider settings, part 5
  • Qualify Frequency: The peer will be contacted this often to check its availability. The default values is 60 seconds, i.e. the peer would be contacted every 60 seconds.
  • Qualify: Sets how often the provider is contacted to check its availability. Choose a short interval if your firewall has a short timeout for UDP sessions (User Datagram Protocol). SIP is using UDP. The unit of time is seconds.
  • NAT: If your network is using Network Address Translation (NAT) and you have problems with one-way audio you might have to change the NAT settings because different factors (providers settings, the SIP phone or router being used, etc.) influence these settings. If you use AskoziaPBX in a public network you might want to disable NAT completely. You can choose between different settings: always use NAT mode, only use NAT mode according to RFC3581 (rport), never attempt NAT mode or RFC3581 support or assume NAT mode but do not send rport.
  • DTMF-Mode: There are different standards used to transmit DTMF (Dual-tone multi-frequency) to SIP providers. Try a different standard when keypresses are not transferred correctly. inband transfers the keystrokes as tones. Make sure the audio codec is of high enough quality. auto, rfc and info transfer the keystrokes within the SIP coding. More about this in Codecs.
  • Session Timer: SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
  • Default Expiry: Defines the default duration in seconds of incoming and outgoing registration.
Advanced SIP provider settings, part 6
  • Authorization User: Some Providers require a separate username for authorization.
  • From User: Some SIP providers differentiate between the username for authorization and the actual username. This can be used in conjunction with outgoing Caller ID specific settings. If this is the case with your provider, enter the correct value here.
  • Disable From User: Activating this option disables the fromuser field in sip.conf. By default, the username of this account is used.
  • From Domain: By default the standard domain name is used. If another name should be used enter it in the text box.
Advanced SIP provider settings, part 7

Further options can be activated by clicking on Expert Options shown in the image below. First, you should save your configurations by clicking on the Save button. A click on Expert Options leads to the GUI options within the Miscellaneous section of the Advanced chapter, where the Enable Expert Options checkbox can be activated.

Expert provider settings notification

The following additional options are now available. These options should only be used by experienced users and only if they are actually required.

  • Manual Dialplan Global: Changes for the global dialplan can be entered here, which then appear in the top of extensions.conf.
  • Manual Dialplan Incoming: Changes for the incoming dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration.
  • Manual Dialplan Outgoing: Changes for the outgoing dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration. As an example, if you want to globally block numbers, the following code can be entered. In this case, all numbers, beginning with 0900 would be blocked.
ExecIf($["${REGEX("^0900[0-9]+" ${EXTENSION_FAILOVER})}" = "1"]?Hangup()) 
  • Manual Attributes: Here, you can directly change Asterisk's configuration files. Manual key-value pairs can be entered in addition to the generated configuration (i.e. configoption=value). These settings will be appended to this item's Asterisk configuration file or context.
Advanced SIP provider settings, part 8, manual attributes

More information about manual attributes can be found in the Manual Attributes section of the Help for Integrators chapter.

Click Save to finish the configuration.


IAX Provider

InterAsterisk eXchange (IAX) protocol is native to Asterisk and supported by a number of phones. Like SIP, it is used for enabling VoIP connections.

General Settings for IAX Providers

The general settings include the Name of the provider and the Host. The Host is either a URL or an IP address. A Username und Password are required to complete the basic provider setup.

From the Language list, the language for the audio prompts for this provider account can be chosen. The specified Public Number will be read back to the caller when reaching voicemail. By default, the account's username is used if it is numeric. If the username is not numeric, the internal extension the call was routed to will be read back. It is advantageous to enter your entire call number in this field to make sure the caller is not just told the extension.

General settings for IAX providers


Advanced Options for IAX Providers

Advanced Options are additional parameters, required by some providers. Most users do not require these options and can ignore them. The following advanced options are available for IAX providers.

  • With Blacklists and Greylists you can prevent specific numbers from calling in. Blacklisted numbers receive the busy signal immediately, grey listed numbers are forwarded to the voicemail of the extension the caller tried to reach. Of course, this only works if voicemail for this phone was configured. Specify one black- or greylisted number per line. Select Blacklist anonymous calls or Greylist anonymous calls if you want all calls with withheld number to be black- or greylisted.
Advanced IAX provider settings, part 1, blacklist and greylist
  • Account code: This code is generated individually for each account. Usually there is no need to change it.
  • Authentication: Two schemes are available. md5 (Message-Digest Algorithm 5) is a widely used cryptographic hash function (encryption method, which is used here to transmit the password in a hashed form). Choose plaintext to transfer the authentication data unencrypted.
  • RSA Keys You can authenticate incoming calls with public RSA keys and outcoming calls with private RSA keys. After activating the desired option, specify the keys in the text boxes below.
  • Registration: In some cases registration is not permitted by a provider. Select Do not register with this provider if this applies to your provider. In most cases AskoziaPBX is able to do the formatting of the registration correctly. Just a few providers expect a special formatting of the registration string. You can specify a registration string in the same format Asterisk uses in the text box.
  • Qualify: Sets how often the provider is contacted to check its availability. Choose a short interval if your firewall has a short timeout for UDP sessions (User Datagram Protocol) as IAX uses UDP. The unit of time is seconds.
Advanced IAX provider settings, part 2

Further options can be activated by clicking on Expert Options shown in the image below. First, you should save your configurations by clicking on the Save button. A click on Expert Options leads to the GUI options within the Miscellaneous section of the Advanced chapter, where the Enable Expert Options checkbox can be activated.

Expert provider settings notification

The following additional options are now available. These options should only be used by experienced users and only if they are actually required.

  • Manual Dialplan Global: Changes for the global dialplan can be entered here, which then appear in the top of extensions.conf.
  • Manual Dialplan Incoming: Changes for the incoming dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration.
  • Manual Dialplan Outgoing: Changes for the incoming dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration. As an example, if you want to globally block numbers, the following code can be entered. In this case, all numbers, beginning with 0900 would be blocked.
ExecIf($["${REGEX("^0900[0-9]+" ${EXTENSION_FAILOVER})}" = "1"]?Hangup()) 
  • Manual Attributes: Here, you can directly change Asterisk's iax.conf file. Manual key-value pairs can be entered in addition to the generated configuration (i.e. configoption=value). These settings will be appended to this item's Asterisk configuration file or context.
Advanced SIP provider settings, part 8, manual attributes

More information about manual attributes can be found in the Manual Attributes section of the Help for Integrators chapter.

Click Save to finish the configuration.


ISDN Provider

The general settings include Name of the provider and the Language for the audio prompts for this account. The Hardware Port is the port to which the ISDN line is physically connected.

General settings for ISDN providers


Regional Settings for ISDN Providers

For ISDN accounts, regional settings need to be specified. This is mandatory information needed for all ISDN provider accounts.

ISDN Regional Settings

Most users use the following settings:

  • Type of Number, the ISDN-level Type Of Number (TON) or numbering plan, used for the dialed number. The list contains "Unknown", "Private ISDN", "Local ISDN", "National ISDN", "International ISDN", "Dynamically selects the appropriate dialplan", and "Same as dynamic (underlying number is not changed)".
  • International Prefix is used when dialing international numbers (usually "00")
  • National Prefix is used when dialing numbers within your country (usually "0")
  • Local Prefix is used when dialing numbers within your city (e.g. "0711")
  • Private Prefix is used when dialing numbers within your company (e.g. "07115678")


Advanced Options for ISDN Providers

Advanced Options are additional parameters, required by some providers. Most users can ignore these. The following parameters are available for ISDN providers.

  • With Blacklists and Greylists you can prevent specific numbers from calling in. Blacklisted numbers receive the busy signal immediately, grey listed numbers are forwarded to the voicemail of the extension the caller tried to reach. Of course, this only works if voicemail for this phone was configured. Specify one black/greylisted number per line. Select Blacklist anonymous calls or Greylist anonymous calls if you want all calls with withheld number to be black/greylisted.
Advanced ISDN provider settings, part 1, blacklist and greylist
  • Account code - this code is generated individually for each account. Usually there is no need to change it.
  • Manual Attributes - here you can directly change Asterisk's isdn.conf file. More information about that in Manual Attributes.

Click Save to finish the configuration.


Analog Provider

The general settings include the Name of the provider and the Language for the audio prompts for this account. The Hardware Port is the port to which the analog line is physically connected.

General settings for analog providers


Advanced Options for Analog Providers

Advanced Options are additional parameters, required by some providers. Most users can ignore these. The following parameters are available for analog providers.

  • With Blacklists and Greylists you can prevent specific numbers from calling in. Blacklisted numbers receive the busy signal immediately, grey listed numbers are forwarded to the voicemail of the extension the caller tried to reach. Of course, this only works if voicemail for this phone was configured. Specify one black/greylisted number per line. Select Blacklist anonymous calls or Greylist anonymous calls if you want all calls with withheld number to be black/greylisted.
Advanced analog provider settings, part 1, blacklist and greylist
  • Answer or Hangup Polarity Switch In some countries, a polarity reversal is used to signal the answer or disconnect of a phone line.
  • Performing Busy Detection can be useful either in an effort to detect hangup or for detecting busies.
  • Account code - this code is generated individually for each account. Usually there is no need to change it.
  • Manual Attributes - here you can directly change Asterisk's analog.conf file. More information about that in Manual Attributes.

Click Save to finish the configuration.


Call Routing

Intelligent Callback is good way of minimizing the limitations caused by only having one line. Whenever a call is coming in via this provider, Askozia checks which extension has last called this number. In case the number is recognized, the call is routed to this extension. If the number is unknown, it is routed to the default extension of this provider. Below you can specify the period of time (in minutes) intelligent callback is valid for.


Outgoing Patterns

There are many different purposes for outgoing patterns. By default, All outgoing calls via this provider is selected. But as soon as you want to use more than one provider, you need to tell Askozia which Provider you would like to use for the current call. For this purpose, Askozia brings a number of pre-defined patterns like Dial 1 as a prefix if you want to use this provider etc, as shown below.

Outgoing patterns, pick list

You can also define your own patterns by selecting I'd like to define my own patterns. Now you can define your own patterns in the text box Outgoing Patterns. If you want to define more than one pattern for a provider, the patterns need to be separated by line breaks. In other words, one pattern per line.

Outgoing patterns, custom patterns


A couple of practical examples are listed below to explain the principle of outgoing patterns.

  • The easiest case is to use one provider for all outgoing calls. To configure this, simply enter X! in the text box. This will match any amount of numbers from 0 to 9. If nothing is entered in the text box only incoming calls can be received but no outgoing calls can be made.
  • Different providers require different patterns. For example, a provider in Germany could ask for numbers beginning with "49" instead of "0", while the users might want to use "0" on their phones. In this case, 49+0|. can be defined as an outgoing pattern. If an outgoing number starts with "0", "0" will be replaced with "49". This allows to dial numbers within Germany as usual. To allow international calls, press enter on your keyboard to start a new pattern in the next line of the text box. For example, 1|. can be entered. In that case, "1" has to be dialed before continuing with the international number you wish to call.
  • If you want to use a certain provider only when dialing a special prefix (e.g. "9") simply enter 9|. in the text box. The "|" character eleminates the "9" and only transfers the following number matched after it to the provider. So, by dialing "91234567", the "9" is matched and discarded because of the "|" character. Then "1234567" is sent on to the provider.
  • If you want to use a certain provider only for local calls, you are forced to enter the area code for each call when not using a dial pattern. If your are code is for example "040" then the pattern to avoid entering the area code is 040+Z.
  • If international calls should be handled by a different provider, this can be automated. To route all calls to Spain (country code 0034) to the provider using the prefix number "01111", the outgoing pattern is 01111+0034. This pattern matches all numbers starting with 0034 and adds 01111 to the front of them before sending them on to the provider.
  • The same method can be used to route calls to mobile phones automatically over a different provider. For German mobile numbers the pattern is 02222+01[5-7]X if the provider's prefix number is "02222". This pattern matches all calls to numbers starting between 0150-0179 and adds 02222 to the front of them before sending them on to the provider.

You can define individual outgoing patterns by using the following character set:

  • + - adds a prefix. The pattern 1+555 matches "555" and passes "1555" to the provider.
  • | - removes a prefix. The pattern 1|555 matches "1555" but only passes "555" to the provider.
  • X matches the digits 0-9
  • Z matches the digits 1-9
  • N matches the digits 2-9
  • [13-5] matches any digit in the brackets (here 1,3,4,5)
  • . matches one or more characters (not allowed before "|" or "+")
  • ! matches zero or more characters (not allowed before "|" or "+")


Emergency Numbers

Emergency numbers for the chosen provider can be entered in the text field shown below.

Emergency Numbers


Incoming Extensions

As with the outgoing patterns, it is necessary to do some minimum configuration otherwise no incoming calls can be received. Askozia add.png adds incoming patterns, Askozia delete.png deletes them.

Incoming extensions

To explain the principle of incoming extensions, here are a couple of practical examples:

  • The easiest use case is to leave the text box empty and to just select one telephone in the drop-down menu as the destination. All incoming calls will then be routed to this telephone. This case is therefore known as catch all.
  • If you click Outgoing Calls Only a drop-down menu opens, showing all available devices, conference rooms and applications. Choosing a destination and assigning an incoming pattern (for example a telephone number) in the text box makes the component available from the outside world. If Outgoing Calls Only remains selected, no incoming calls can be received from this provider. If a device is chosen but no incoming pattern specified, all incoming calls are routed to this telephone.
  • If a device is chosen and a number is entered in the the text box next to it (e. g. 432121), this routes all calls to this number through this device.
  • If more than one phone is configured with the same incoming pattern, they will all ring at the same time when there is an incoming call. This is an easy way of defining call groups.

You can define individual incoming extension patterns by using the following character set:

  • X matches the digits 0-9
  • Z matches the digits 1-9
  • N matches the digits 2-9
  • [13-5] matches any digit in the brackets (here 1,3,4,5)
  • . matches one or more characters (not allowed before "|" or "+")
  • ! matches zero or more characters (not allowed before "|" or "+")


Integrator Panel
Auflistung der abgewiesenen Nummern


Automatic Fax Detection

Especially for the users of analog and ISDN lines, automatic fax detection is useful because they do not have to reserve one number only for the fax (see Incoming Extensions for details). In this case, Askozia checks every incoming call for fax content. This causes a two second delay for all incoming calls. If a fax is recognized, Askozia forwards it to the appropriate device. Apart from the delay, incoming calls are not influenced by automatic fax detection.

A click on the drop-down menu opens a list of available faxes, as long as at least one fax accounts exists.

Automatic Fax Detection


Extend Provider

To automatically switch to a different provider, in case all lines of one provider are busy, you can select an alternative provider.

Provider list
Provider settings and dial pattern

The failover feature is still available as a sub option within the new extend provider feature. Select play tone if you like to hear a tone signal before every call once you are not using the default/first provider anymore. When updating from 2.2.2 this option is automatically selected.

Alternative provider and signal tone check box


Caller ID Options

This section is about adjusting the Outgoing Presentation and the Incoming Modification. The external presentation is the text seen by the person receiving the call on the display of the telephone (e.g. "Bob from Askozia"). The need for incoming modification is less straight forward to understand. More about that in the section below.


Outgoing Presentation

By default the outgoing presentation is set to the caller ID of the telephone placing the call. Select the option in the middle to use the default caller ID. It is also possible to transfer a user-defined text (e.g. a name) or a call number. Choose always send Caller ID above and enter the text in the text box. It is also possible to transfer the caller ID (in this case a phone number) plus an appended area code. Choose the bottom most option and enter the area code and the number in the text box below.

Outgoing caller ID presentation


Incoming Modification

You could easily assume that it is always desirable to see the caller's original caller ID on the telephone display. But there are cases where it needs to be modified. For example, when different numbers are routed to the same telephone. This might happen when multiple companies are sharing one telephone system. In this case, it is interesting for the user to see which number the caller has dialed to answer the call with the correct company name. Select Replace Caller ID with text above for this case.

Incoming caller ID presentation


Codecs

AskoziaPBX supports a variety of free and licensed Codecs for both audio and video as shown below. Codecs are explained in detail in Help for Integrators for experienced users.

Audio codecs
Video codecs


DECT bases

A new DECT base station can be added by clicking on Accounts add DECT base station eng.png on top of the account overview for devices.

An account for the DECT base needs to be set up before you can add DECT handsets. Afterwards, DECT handsets can be added by setting up IP phone accounts.

The settings below are required for all supported DECT bases.

  • Model: Select the DECT base of a supported brand that you want to provision with AskoziaPBX. Please check if the firmware named here is the same as for your DECT base.
  • MAC Address: Enter the MAC address of the DECT base you want to provision with AskoziaPBX. It can usually be found on the back side of your DECT base.
  • Name: Enter a name for the DECT base that is used for your reference.
  • Language: Audio prompts will be played in the selected language for this DECT base account. The selected language will also be used for the Call Control CTI web interface for this account.
  • Password: The password that is required to log into the web interface of this DECT base.
General account settings for DECT base stations


Beside the standard settings, further settings may appear depending on the selected DECT base model.

  • PIN: The PIN is typically used to register the DECT handsets with the DECT base. Each PIN needs to consist of 4 digits.
PIN for DECT base stations


Click on Save to finish and save the configuration of a DECT base. After saving your configuration, the DECT base appears in the DECT Bases account overview of the Device Accounts menu. Furthermore, the Handsets column reveals, how many DECT handsets can be connected to each DECT base.

Device accounts, DECT bases


To assign specific numbers to single DECT handsets, the web interface of the corresponding DECT base needs to be accessed which can be done by clicking on Askozia open device web interface.png.

In any case, it is necessary to read the manual of the DECT base manufacturer. Before connecting DECT handsets to AskoziaPBX, they need to be reset to factory defaults. To connect a DECT phone to AskoziaPBX, an IP phone account needs to be created. In the Auto-Configuration settings of the account, the desired model and the DECT base need to be selected.


Gigaset N510

For Gigaset N510 IP PRO DECT bases, PIN and password are identical and both to be defined in the password field as a 4-digit number. The PIN is used to log into the web interface of the DECT base and to register the DECT handsets with the base.

Enter the IPUI address in the auto-configuration settings of the phone account of the DECT handset to bind the handset to this account.


Gigaset N720

For Gigaset N720 DECT IP Multicell bases, PIN and password are separated. The password is used to log into the web interface of the DECT manager. As the PIN is assigned automatically by the DECT base, you need to enter the web interface to access it.

Gigaset N720 DECT IP Multicell bases can be organized into several centrally managed clusters, each with it’s own name and MAC address. A base station always synchronizes itself with a base station that has a higher sync level. A Sync level of '1' is the highest possible sync level and can only appear once for each cluster. Add a new cluster by clicking on Askozia add.png. Delete the last cluster by clicking on Askozia delete.png.

At least one cluster needs to be specified to make calls with the Gigaset N720 DECT IP Multicell base. For this purpose, the MAC addresses of both DECT base and manager need to be entered.

Specifying clusters for DECT base stations


Enter the IPUI address in the auto-configuration settings of the phone account of the DECT handset to bind the handset to this account.

Snom M300 + M700

User name and password are required to register at the web interface of the DECT base. The password is specified in AskoziaPBX as explained above. The user name is "admin".

To register a DECT handset with a DECT base, select register and enter the PIN specified in the account settings of the DECT base. Enter the IPEI address in the auto-configuration settings of the phone account of the DECT handset to bind the handset to this account.


Yealink W52P

User name and password are required to register at the web interface of the DECT base. The password is specified in AskoziaPBX as explained above. The user name is "admin".

To register a DECT handset with a DECT base, push the registration key of the DECT base. Afterwards, start the registration at the DECT handset and enter the PIN specified in the account settings of the DECT base in Askozia's web interface.


Phones

Phone accounts are sorted by technology type (SIP, IAX, ISDN, analog and external). If your PBX hardware includes analog ports, analog phone accounts are automatically created.

Askozia bullet green.png in front of the extension of a VoIP telephone indicates that the phone is registered with and reachable by AskoziaPBX.

Active accounts have a status (red, green or gray). Account active.png

Inactive accounts do not have a status. Account inactive.png

SIP Telephones Overview
External Telephones Overview
Analog Telephones Overview


Add one or more phones

To add a telephone, click Phones in the menu bar. Choose the type of the telephone (SIP, IAX, ISDN, analog or external) to get into its specific configuration menu. An external phone account connects an internal extension to an external number such as a mobile phone. External phones can not be added until at least one provider account was created. External phone numbers do not require the specification of outgoing patterns.

Types of Telephones


Edit one or more phones

To edit one phone account, just click on the desired extension to open the accounts options.

Open options for a single account

To edit multiple accounts at the same time, mark the desired accounts via the check boxes in the overview. Several symbols will appear in the upper right corner of the overview.

Mark accounts for editing

Phone edit 01.png - Edit selected phone(s)

Phone edit 02.png - Delete selected phones(s)

Phone edit 03.png - Go to phone's webinterface

Phone edit 04.png - Restart selected phone(s)

Phone edit 05.png - Enable/Disable selected phone(s)

Phone edit 06.png - Delete voicemail of selected phone(s)

Specifics for editing multiple phones

The available options differ slightly, depending if a single phone or a group of phones is selected. That’s because not all options make sense in group edit mode.

Options which cannot be configured parallel

Fields with different configurations inside the single accounts are marked with multiple values. If wrong changes have been made, the configuration can be reverted back to the previous settings, unless these changes weren't saved already.

Multiple values text.png text fields

Multiple values checkbox.png check boxes

Multiple values dropdown.png drop Down Menus

Multiple values conffields.png configuration areas (i.e. codecs)


Automatic IP phone detection

Automatically detect VoIP phones in your local network (LAN) as an alternative to Auto-Configuration. This is very comfortable as you don't have to enter your phone's MAC address.

Automatic Phone Detection

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 normal Auto-Configuration and enter its MAC address manually.


General Settings

The general settings allow the basic configuration of a phone.

  • In the text box Number, the extension of the telephone is specified.
  • The caller ID is the name or number that dialog partners see on their phone displays.
  • Default Provider determines one provider this phone uses for external calls. Please note that this overwrites your outgoing dial patterns.
  • 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.
General Settings
  • Ring Length is 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.

  • Description gives you the opportunity to add an additional description for the phone.

This description is ignored by AskoziaPBX but may be handy to clarify a phone's particular purpose or location.


Security

  • In order for the phone to register with AskoziaPBX, a secure password is generated automatically for SIP and IAX phone accounts. 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. This function is available for all telephone types, conference rooms and applications. 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.

Phone security settings
  • For IAX phones, an authentication method can be choosen from the drop-down list. Askozia supports plaintext and md5. plaintext is least secure as it sends passwords cleartext across the net. md5 on the other hand uses a challenge/response md5 sum arrangement. Still, md5 requires both ends to have plain text access to the password.


Phone Connectivity

For using external phones with AskoziaPBX, the provider to be used and the external number of the phone have to be specified. Please note that the external number do not need an outgoing pattern. Even if one is configured for the desired provider.

Connectivity settings for external phones


Auto-Configuration

The configuration menu of SIP phones provides the additional menu item Auto-Configuration. By entering a phone's MAC address it can automatically be configured with the same settings specified in its AskoziaPBX phone account.

You can either find the MAC address of your phone on its under side or through the web interface of the phone. To gain access to the web interface, first push the Help button on your SNOM phone.

Auto-Configuration

Some phones require a URL for auto-configuration. You always find the current manufacturer specific URL below the drop-down menu.

In the section Speed Dial you can assign numbers to the speed dial buttons and busy lamp fields (BLF) of your phone. Please make sure that you only assign as many numbers as your phone has speed dial buttons.

After entering the information, click Save. Now, restart your phone by disconnecting it from the power supply for a moment. After restarting, your phone is configured and registered with AskoziaPBX.

Supported Phones

AskoziaPBX supports auto-provisioning for a variety of IP phones. Details about the supported models are provided in the Supported Hardware chapter.


Provisioning via VPN

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

Link to the provisioning file

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

snom Webinterface

-> The phone will reboot and download the XML file.

Yealink Webinterface


How to configure a soft phone

What configuration is required to use a soft phone with AskoziaPBX is shown here.

To find examples for softphones please have a look here.


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.

Call Control CTI settings for phone accounts

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.

Call Control CTI user instructions

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.

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.

Client user interface settings for phone accounts

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.

Askozia landing page


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

If you want to receive notification e-mails for missed calls, select notify me of missed calls via e-mail. Furthermore, you need to enter the e-mail address to which the notifications should be sent to in the text box below.

To enable voicemail, select enable voicemail to e-mail and enter an e-mail address. Once enabled, all incoming voicemail messages will be send 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.

When choosing a phone or a call flow in forward on timeout, the call will be forwarded to the chosen extension when the ring length expired.

Forward on busy plays the busy signal or forwards an incoming call to the chosen extension, if the called phone is busy.

Call Notifications and Voicemail, first part
  • 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.
Call Notifications and Voicemail, second part


Outgoing Caller ID

The outgoing caller ID of each phone can be set for each provider. Leave the field blank to use a provider's default settings. Use the formatting "Your Name <number>" as shown in the screenshot below.

Outgoing Caller IDs

"49" is the country code for Germany. "5331" is the area code. Instead of the "x" letters, the desired number has to be used. For some providers, the country and area code may have to be used 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.

Parallel call and secretary feature configuration


Codecs

Codecs are explained in Help for Integrators for experienced users.

Audio codecs
Video codecs


Call Recording

To activate and store call recording, storage needs to be set up first in the System settings.

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.

Call Recording Options for single phones


Manual Call Recording

Activated 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.



The following video tutorial shows how to use manual call recording.



Automatic Call Recording

To record all phone calls, select Record all calls in the call recording settings of the Miscellaneous section in the Advanced menu. This can be done globally for all phones or for individual phones in their respective accounts. To activate recordings for individual phones, first make sure that Enable global call recording in Miscellaneous is activated. Afterwards, configure the account of the phone.


Hot Desking

Hot Desking is used in companies with shared office infrastructure. When desks are 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 that possible.

To activate Hot Desking in AskoziaPBX, the extension settings have to be opened. In the submenu Hot Desking the check box "activate Hot Desking" has to be set. Further, a password for the account has to be set.

After configuring the settings, the following starcode has to be entered on the desired phone.

*0*XXX (Extension of the Hot Desking Account)

Now it's required to enter the set password and confirm it by pressing the pound key.

Then, the telephone will reboot and load the desired account. If the desired account have been active on another phone, before activating, the accounts will switch on the phones.

With *0 accounts can be deactivated on phones.

General settings Hot Desking


Advanced Options for Phones

Advanced Options are additional parameters not required in most cases. The following parameters are available.

Advanced options for phones
  • 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. This option is available for SIP phones.
  • 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. This option is available for SIP phones.
  • 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
  • Night Switch Exception makes the phone available from the outside, even when the night switch is activated. This option is available for all phones.
  • Phone Book Exception makes this phone unavailable in the local phone book which is transfered to all auto-configured phones registered with Askozia. This option is available for all phones.
  • 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.
  • With SIP Permit/Deny you can limit the SIP traffic to a certain IP range. This option is available for SIP phones.
  • NAT offers several NAT modes. Choose the required mode from the drop-down menu. This option is available for SIP phones.
  • DTMF Mode specifies how DTMF signals are transfered via SIP. Choose the required mode from the drop-down menu. This option is available for SIP phones.
  • Busy Level defines how many parallel calls the phone can handle before signaling "busy". This option is available for SIP phones.


Fax

To add a fax, click Fax in the menu bar. You can use analog and virtual faxes with Askozia. A virtual fax converts incoming faxes to PDFs and sends them to your e-mail address. Choose Analog or Virtual to add a new fax.

Fax types
Fax overview


Analog Fax

The internal number of a fax is assigned in the text box Number. The Caller ID is the name or number the caller sees on the fax display. For analog faxes, the Hardware Port needs to be set. The hardware port is the physical hardware port this fax is connected to.

General settings for analog fax


Advanced Options for Analog Fax

The advanced options for analog fax configuration allow activating polarity switch and busy detection.

Advanced options for analog fax

In some countries, a polarity reversal is used to signal the answer or disconnect of a phone line. Performing busy detection can be useful either for detecting hangups or for detecting busy lines.

Click Save to finish the configuration.


Email to Fax

A virtual fax allows you to send and receive faxes without using a fax machine. The features of a virtual fax are also called email-to-fax and fax-to-email. When using a virtual fax, incoming faxes are automatically converted into PDF files and send to you via email. To send a fax, simply send an email to your virtual fax. The virtual fax converts text and PDF attachments into the fax format and faxes them. In the following section we explain how to set up a virtual fax in AskoziaPBX.


Configuration of a virtual fax

The number of a fax is assigned in the text box Number. The Caller ID is the name or number the callee sees on the fax display. Virtual faxes require an E-mail address to which the faxes should be sent.

General settings for incoming virtual fax

Default provider is the standard provider used for this virtual fax. Block providers allows you to exclude certain providers for faxing (e.g. SIP providers without T.38 support). Select the page format common in your country. Selectable in the the drop down menu are A4, letter, legal, B4 and A3. The local station ID is the callerID used by this virtual fax. Fax header information allows you to add further information. Retries defines the number of retries if the target fax is busy. Email size defines the maximum email size in megabytes. Keep in mind that large attachments require more CPU time. This is especially important on embedded systems with limited resources (e.g. Askozia Small Business Appliances). Maximum number of pages prevents faxes from exceeding a certain number of pages from being send.

To be able to send faxes, you need to configure the POP3 settings of your email address for your virtual fax. This is necessary because the virtual fax is downloading the emails from the email inbox defined here, before sending them as a fax.

General settings for outgoing virtual fax
General settings for outgoing virtual fax

In the advanced options, error correction and T.38 gateway functionality can be activated, if supported by your provider. Optionally, you can set an activity timeout in seconds. In advanced options, you can also define a standard transfer rate as well as a minimum and a maximum transfer rate.


Advanced settings for outgoing virtual fax

Click Save to finish the configuration.

To send a fax with your virtual fax, simply send an email to your virtual fax. AskoziaPBX converts text and PDF attachments into the fax format and faxes your email. Open your email client. Enter the email address of your virtual fax in the TO field. As the subject enter the fax number, where you’d like to send the fax to. Enter the text you’d like to fax in the email body and/or attached a PDF file. Supported are plain text and HTML emails. PDF attachments need to be in the format standard to fax machines in your country. When sending an email with text and a PDF attachment, a new page is started for the content of the attachment.


Fax Configurations

Here we listed tested configurations for multiple email providers.

Strato

configuration for Strato accounts

GMail

configuration for GMail accounts

GMX

configuration for GMX accounts

Web.de

configuration for web.de accounts

Hotmail

configuration for Hotmail accounts
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