To access Domain Templates, from the domain level, click on Advanced and Customize

From this page, you can use the drop-down fields to choose the type of template and page you want to customize:

Admin Domain pages

User Portal

Email templates

Phone PnP templates

Once you have chosen the page, the HTML code will appear in the field below.

For more information on editing, see https://vodia.document360.io/v1/docs/appearance.
Once editing is complete, click Save and your changes will be applied.

Cell phones are widely available for most users and can be integrated with the Vodia PBX in two ways. As a first possibility, users can install the Vodia apps on their devices and use the data network to connect with the PBX, which is described in a seperate document.

As an alternative way, cell phones can be used through their PSTN calling features. This has the advantage that the user does not have to install an app and that it tends to work better when data connectivity is not as good as telephony cell phone service. Similar to SIP VoIP phones, they can be used to receive and make phone calls. However the call is not routed natively routes through a SIP network, the cell phone typically uses a mobile phone standard like GSM or CDMA.

The settings for the cell phone allow you to configure one or more cell phones to your extension. Cell phone integration is different from having two phones registered to the same extension. Registering two phones to the same extension allows you to merely receive calls at one of two locations, while cell phone integration allows you to receive calls at any location, as well as be reached at numerous phone numbers. Cell phone configuration allows you to be accessible at all times, even in the event of a power failure. If the office extension (or remote extension) is disconnected from the system, calls will automatically be sent to your cell phone. Once the phone is reconnected, incoming calls will automatically be rerouted to the extension.

Settings

Cell phone number: In this setting the user can put one or more cell phone numbers. If there is more than one number, spaces must be used to seperate the numbers. While most users populate this field with a cell phone number, actually any phone number can be used (for example land line numbers). The numbers must be entered without any spaces or hyphens (e.g., 9781234567).
Cell phone number (inbound only): The user may specify additional cell phone numbers that are used for detecting inbound calls from the user; however those numbers are not used for outbound calls. This is useful to simplify authentication for call from the users associated numbers without disturbing the user on those numbers when locating the user.
Numbers that will never get to the cell phone: Because cell phones are usually carried as a personal item during the whole day, it is important to define who can call the cell phone and who cannot (as well as the times). This settings explicity blocks certain numbers from calling the users cell phone. The setting contains a list of numbers and wildcards, seperated by space. For example, the list "400 5*" would block extension 400 and all extensions starting with 5 to call the cell phone.
Numbers that will get to the cell phone anytime: Similar to the previous setting, this list defines the important contacts that can call the cell phone at any time, regardless of the time restrictions below.
Offer special menu when call is answered on cell phone: This setting provides users with additional menu options when answering a call from the cell phone. These options are detailed in the following diagram.

• If the user presses "1" the call gets connected to the cell phone. Until then, the caller will continue to hear the ringback tone.
• If the user presses "2" the call gets rejected. The PBX will then apply the rules when the VoIP phone would send a busy signal.
• Upon pressing "3" the PBX will send the caller to voicemail.
• When pressing "4" the system will prompt for the number to redirect to. The number must be terminated with the pound kex ("#").
• When pressing "5" the call gets parked, as if the user accepts the call and then dials the park star code.
• When pressing "6" the call is added to the list of callback numbers. This is useful to stack up numbers that can be called back, e.g. on the way home from the car.
• The digit "0" can be used if the operator account has been assigned to the domain. This is typically an attendant that takes cares about incoming phone calls for the domain.

When this cell phone calls the PBX: This setting defines what the system should do on incoming calls. It can either offer the personal virtual assistant or just apply the general rule for routing inbound calls. In case the private virtual assistant is enabled, the PBX offers the following options:

• 0: If there are one or more parked calls, the option "0" will resume call first call.
• 1: By pressing "1" the caller can start a new outbound call.
• 2: If there are no parked calls, ask the system for a callback. This is important when calls from the PBX to the cell phones are significantly cheaper than the other direction or when the user is using his private cell phone for business purposes.
• 2: If there is one parked call, offer the possibility to transfer the call.
• 3: If there are no parked calls, go the the mailbox.
• 3: If there are more than one parked call, offer the possibility to put the calls into a conference.
• 4: If there are no parked calls, go to an auto attendant in the domain. This can be used to dial internal numbers.
• 4: If there are two parked calls, offer the possibility to connect the two parked calls.
• 5: Call the next number on the callback list.
• 6: If the account has a credit for outbound calls, it can be retrieved with this option.

Initiate call back after hangup: In order to avoid charges to the calling cell phone, the PBX can automatically call back the cell phone after it called into the PBX. This feature is useful when employees have to pay privately for cell phone minutes. With this feature, those charges can be avoided.
Include the cell phone in calls to extension: When someone calls the extension directly (which means not through a hunt group or agent group), the PBX can include the cell phone not at all, immediately or after a certain time. This makes it possible to start the cell phone call only if the call was not picked up on the VoIP phone. If the VoIP phone is not registered, the PBX will not wait until the time has elapsed.
When calling the extension in a hunt group: When a call hits a hunt group, the PBX can include the cell phone in the group call. In this case there is no option to delay the call, however the user can include the cell phone when the VoIP phone is not registered. This is a great option for hosted PBX when there is a risk that an office gets temporarily offline when cell phones can take over.
When calling the extension in an agent group: Similar to the previous setting, the cell phone can also be included in agent group calls.
Include cell phone for web callback: When using the click to dial feature from the web browser, this setting controls if the cell phone should also be included in the callback. As with the settings before, the callback can be made depending on the registration of the VoIP phone.
Specify time when system calls the cell phone: If the user wants to limit the cell phone inclusion to a certain time of day, he can specify this in this setting. If service flags have been set up in the domain, they can be used for that purpose; otherwise the user can just specify the times in the fields.
When this cell phone calls the PBX (VPA) The steps to setup the Virtual Private Assistant:

• The incoming calls to the PBX should land directly on the Auto-attendant.
• The cell phone number of the extension should be entered in the "redirection" section in the same format that it comes in with, so that the PBX can match and recognize that the incoming number and the cellphone number tied to the extension are identical.
• Under Domain > Settings > General Settings , turn on "Trust caller ID" to ON.
• This extension should have a PIN number assigned to it (More than 4 digits is usually accepted).

  Web Attendant Console

  The web attendant console lets you manage calls from your user portal on the PBX, like hang up, blind transfer, hold, attended transfer or making a call etc. All these actions can be performed very easily by either clicking or dragging and dropping as explained below. It also lets you monitor the other extensions on your domain, like whether they are registered, in a call or idle etc. For more on logging in to your user portal, go to Logging In .

  Extension Monitoring

  On the home page of your User Portal you will see the status of all other extensions (and other accounts like Agent Groups and Hunt Groups) in your domain.

  • Orange dot next to the extension means it is not registered with a physical desktop phone or not registered at all.
  • Green dot next to the extension means it is registered and is idle.
  • Orange circle over the extension means it is registered but is busy in a call.

  

  Call Controls

  When a user is logged in to his User Portal, the calls are shown on his portal’s home page along with call controls which are used for call management. There are three sections for the calls:

  • Ringing Calls (for calls that are currently ringing and can be picked up).
  • Active Call (the call to which the user is talking at the moment)
  • Calls On Hold (calls that are on hold)

  1. Ringing Calls

  When a new call comes in and the registered phone rings, the call is also shown in the "Ringing Calls" list. As more calls come in, they keep adding to this list. You have two options for each call: pick it up by clicking on the green handset icon next to the call or hang up by clicking on the cross next to the call. If you hang up, the call will be rejected and disappear from your phone and portal. If you pick it up and the phone supports it, then the call will be connected through your phone (handset, headset or speaker) and the call will move from the ringing list to the "Active Call" list. If there was already another active call, that call will go on hold and move from the "Active Call" section to the "Calls On Hold" list while the new call will become active and move from the "Ringings Calls" list to the "Active Call" section.

  

  2. Active Call

  The call to which you are currently connected and talking to are shown here. You have 4 options:

  (i) Hang Up

  To hang up the call, simply click on the cross beside the call that you want to hang up.

  

  (ii) Hold

  To hold a call, simply click on pause icon (handset with the pause sign) beside the call you want to hold. The phone will go on hold and the call will move from the active call section to the "Calls On Hold" list. All calls that are held will keep adding to the hold list.

  

  (iii) Blind Transfer

  To Blind Transfer a call, simply click and drag the transfer icon (handset with an arrow) beside the call you want to transfer, and drop it on the extension (shown in the monitoring section) you want the call to be transferred to. The call will be transferred and disappear from your portal.
  

  (iv) Attended Transfer

  For Attended Transfer, you will usually have at least one call on hold as you dial and talk to another extension for a possible transfer. The active call will be in the "Active Call" section and the held calls will be in the "Calls On Hold" list. You can now transfer the active call (after telling him/her that there is call for him/her) by dragging the transfer icon of the active call and dropping it on one of the held calls (in the hold list) to which you want the call transferred.

  Step 1:

  

  Step 2:

  

  3. Calls On Hold

  One can either "Resume", "Transfer" or "End" the held call from the option under the burger sign (3 dots on the right hand side).

  

  Making a Call

  To make a call from your phone on which your extension is registered, simply enter the number you want to call and press the Call button or hit enter on the keyboard.
  

  Agent Group Information

  For each agent group the extension belongs to, the WAC shows information about the group. The following information is available:

  Calls

  This sections shows the currently active calls in the group. When calls enter or leave the group, the browser updates the information in real time. The Time shows the time when the call entered the queue. The From field shows the caller-ID of the caller. If the call is connected, the Agent field shows the agent that is talking to caller.

  Call Statistics

  This section provides general information about the number of calls in this queue of the day.

  • The Calls field shows how many calls have hit the group so far.
  • The Connected field counts how many calls actually connected to an agent.
  • The Hang up while waiting counter shows how many calls got disconnected because the caller hung up while waiting and the ACD was playing music on hold.
  • The Hang up while ringing counter shows how many calls got disconnected because the caller hung up while an agent was already alerted and the caller heared ringback tone (unless something else was programmed).
  • If there was a redirection action programmed for the group, the Waiting redirect counter shows how many times a caller was redirected while waiting and the ACD was playing music on hold.
  • Accordingly, the Ringing redirect counter shows how many times a caller was redirected while an agent was already alerted and the caller heared ringback tone.
  • The Key exit counter determines how many times a user decided to exit the group by pressing a button, if this was set up.
  • Go to Groups (under /dom_groups.htm on the PBX) and set the permissions as shown:

  

  And here you can assign the extension you want to monitor the ACD details of, which may or may not be a part of the ACD. The results will look like below.

  

  Duration

  The duration section analyzes how long calls were in their respeoctive states. The information is shown in total (sum of all calls of the day) and as average (sum divided by number of calls). In case that the view shows multiple groups, each group has its own row.

  • In Idle state shows how long there was no call in the group.
  • If the state is Waiting the caller hears music on hold and possibly annoucements, if they had been recorded for the group.
  • The state Ringing is when the caller hears ringback and the agent gets alerted.
  • The state Connected is reached when the caller talkes to the agent.
  • The state Hold means that the caller is connected to an agent, but currently put on hold, for example because the agent is performing a consultative call. Because the hold state is part of the connected state, the duration for this state is shown in brackets.

  Agents

  This section shows for each agent activity information. The information is shown in total and on average per call.

  • The Name field shows the text name of the agent and the extension number.
  • The Idle field shows how much many horus and minutes the agent was idle between calls. The idle time is counted not only within the group, but within the system. That means when an agent performs other calls for other groups or just regular calls, the idle counting starts at the end of those calls.
  • The Connected time shows how long the agent was talking. This field shows only the talking time within the group (if a group has been selected in the web interface).
  • The Hold shows how much out of the connected time the call was on hold.
  • If the agent logs out, this will be reflected in the Breaks field. It shows the intervals when the agent was not available for the group and why. Possible reasons are DND, call forward all and not logged in.

  Call History

  The call history shows the last 20 calls in the agent group. In addition to the fields for the current calls, the history also shows how long the call was connected.

  SIP Trunks

  

  Trunks are used to route calls in and out of the phone system. The term "trunk" originates from old PBX systems, where the PBX was connected to the public telephone network using physical wires called trunks.

  Today trunks transport the conversation over the network. Calls can end up on a PSTN gateway that is located close to the PBX server, but they can also end up on a service provider's network that takes care about the PSTN termination as a service.

  Trunks may have several lines. These lines represent the physical cables that were used in the PSTN world and they are called "CO-lines" in the context of the PBX. A CO line can have one call appearance. CO lines are mostly not used any more and for new installations not recommended.

  There are different trunk types:

  • When using a PSTN gateway, the gateway is typically programmed to send all inbound calls to the PBX. The PBX is programmed to send the outbound calls to the gateway. These trunks are called "gateway" trunks.
  • Many service providers started their business offering PSTN call termination for soft phones running on PC endpoints. This model is mostly used for today's SIP service providers where the PBX needs to register itself to the service provider like a soft phone. This type of trunks are called registration trunks.
  • A new, evolving trunk type is using the ability of modern web browsers to terminate phone calls. These trunks are called WebRTC trunks.
  • There are special trunks that use OPTIONS instead of REGISTER which are not very common but neccessary in some cases.

  The PBX may have several trunks (depending on the license). For inbound calls, the PBX needs to identify the trunk first and then determine where to send the call. This is done using the settings of the trunk.

  For outbound calls, the PBX first checks if the dialed number is an internal resource, for example an extension or an auto attendant. If that is not the case, the PBX needs to select the dial plan that will tell the PBX on which trunk the call should be terminated.

  • If the dial plan has been explicitly assigned to the account that is initiating the call, the PBX will use that dial plan.
  • Otherwise the PBX will choose the default dial plan of the domain where the call comes from.

  The PBX will process the dial plan and then determine the trunk that will be used for placing the call. The headers settings of the trunk will govern the way the call is represented to the PSTN gateway or the service provider.

  Creating a Trunk

  Go to your selected domain in the Vodia PBX interface. You will see the current list of SIP trunks.

  

  The list shows you the name of the trunk, the account that was used in the trunk (this can be empty), the outbound proxy that was assigned to the trunk, the status including the remaining duration of the registration where applicable, the trunk type, the address association, the ANI was was assigned to the trunk and if the trunk was set up as a global trunk.

  On this page, you can also delete, temporarily disable or re-enable trunks. You can also explicity trigger a re-registration of the trunk. In order to create a new trunk, click on the Add button. A pop up window would give you a chance to enter the details in a modal window.

  

  If your provider is in the drop down, the dialog will ask you for the information needed to set up this trunk. This is usually only the username and password, but in some cases you need to add a second username or domain names. If certain phone numbers are associated with the trunk, you can enter those numbers in the creation dialog. The numbers should be seperated by spaces. If you have a range of numbers, you can add them using a hypen as shown in the example. This will set up your trunk with a reasonable configuration. Hoewever you can still edit the trunk and modify the settings.

  A setting that you might want to change is the destination for the trunk. By default the system will try to find a reasonable destination. However if you want to explicitly control this, you can do this in the trunk settings.

  If your trunk provider is not in the dropdown, you can just create a new trunk (use "Other" as provider). This might take a little time to figure out the settings of the trunk. If you feel that your provider should be in the drop down please feel free to contact support to add the trunk.

  Dial Plans

  Dial plans are used to route (and optionally modify) specific dialed numbers to specific trunks. Dial plans are used when an internal destination could not be found.

  Basically, it first compares a pattern with the dialed number according to matching rules. If it matches, it replaces the dialed number according to the replacement rules (if replacement field is not empty) and sends the call to the trunk set for that rule (pattern). If it doesn’t match, it moves on to the next rule, and so on.

  The rules are defined in separate lines and each contains at least:

  • A trunk to which the call is sent if the pattern matches.
  • A pattern which the dialed number must match for the rule to trigger. For example a pattern 9* would mean any dialed number starting with 9 would be sent to the trunk set in this rule.
  • Optionally, a replacement field which replaces the dialed number (according to replacement rules) before sending it to the set trunk. So, for example, if the pattern is 9* and the replacement rule is 1*, then the number 92121234567 will be replaced with 12121234567 . If the replacement field is empty, then the dialed number is not touched as it is sent to the set trunk.

  The dial plan is sorted by the priorities before execution. Lower numbers will be matched first. If the numbers are the same and the dial plan entry is resumed later (e.g. because of a trunk failover), it will continue with the next higher number.

  Dial plans can be global. This means that all domains can use this dial plan. This makes sense in multi-tenant environments where all domains can share the same dial plan and the same trunks.

  The PBX can check if a number is on a do-not-call list. This uses the address book of the user and domain—if the entry has the do-not-call flag set it will not call the number.

  The numbers that are put into the dial plan are usually in the readable format. This means the number is presented in the way that a user would read it. This means that in North America, domestic numbers are 10-digits (without the leading 1) and international numbers start with 011. For the rest of the world, domestic numbers start with a single 0, and international numbers start with a 00. If an area code is present, the PBX will use the local presentation if the call is within the area code. You can change this behavior in the dial plan, for example if you prefer the E164 format in the dial plan.

  Creating a Dial Plan

  

  • Go to your selected domain in the Vodia PBX web interface.
  • Click on Dial Plans under Features as shown above.
  • Give a Name for the new dial plan you want to create.
  • Press Create .

  The new dial plan will appear in the list above. You can now click it in order to configure it. You can also enter any other dial plan in the list, for modification, by clicking it.

  Configuring a Dial Plan

  
  Go to your selected domain in the Vodia PBX web interface.

  • Click on Dial Plans under Features as shown above.
  • Click on the dial plan on the list that you want to configure.
  • Here you can modify the fields for the different rules of the dial plan.
  • Here you can also create new rules or remove rules or enable or disable them.
  • To create a new rule, simply enter all its fields, like the pref number, selecting the rule's trunk from the dropdown, the pattern to match, and the replacement (which should be left blank if you don't want to touch the dialed number before passing it to the trunk). You can also enter a service flag for the dial plan to be active or not, depending on the timings given in the service flag.
  • Press Save.

  The new rule will appear on a separate line on the list of rules.

  Dial plans have a few parameters that control how the system processes the dial plan:

  Name: This is the human readable name for the dial plan.
  Global: If this flag is turned on, the dial plan will be available to all tenants of the system.
  Number presentation in dial plan: This parameter determines how numbers are fed into the dial plan. By default, the system reformats the number that a user has entered into a unique presentation, for example it does not make a different if a user dials 16173998147, 6173998147 or +16173998147.
  Check if number is on do-not-call list: When this flag is enabled the system will first check if a number is on the do-not-call list in the tenant before processing the number.
  Star codes that are handled in dial plan: This setting lists the star codes that are handled by this dial plan. Star codes must start with *. If the list is empty, the dial plan will not handle star codes. Otherwise, the list contains the star codes that are handled by this dial plan, for example when the list is *54 *55 this dial plan will handle *54 and *55, all other star codes will be processsed internally by the system. If a star code ends with a *, the system will match only the beginning. For example when the entry is *54* the system will match *54, *541 and *54123. If the pattern is *, all star codes will be handled by the dial plan.
  Route internal calls: When this flag is enabled, the system will check if the dialed number is an internal resource of the tenant and send the call there without processing the dial plan.

  Dial Plan Entries

  There are several fields in the dial plan:

  Preference. The preference is used to tell the system how to sort the dial plan.
  Trunk. The trunk drop down field contains the trunk that should be used. There are some special fields that are explained below.
  C. This fields stands for CMC, or client matter code. When checked, the PBX will attempt to collect the client matter code from the user before placing an outbound call. This can be useful for accounting purposes.
  P. When this field is turned on, the PBX will ask the user to enter the PIN code. The pin code can be set under each extension level > Account Profile > PIN setting. This can be useful to protect expensive dial plan entries from abuse, e.g. employees walking into office calling international destinations.
  Pattern. The pattern field defines when the entry matches. This field is described below.
  Replacement. When the pattern matches, the PBX may transform the user input, for example put numbers in front of the dialed number. This is also explained in more detail below.
  Service Flag. Each dial plan entry may be enabled or disabled with a service flag. That makes it possible to enable certain entries based on time or based on manually set flags.
  Status. In addition to the service flag, each entry may be manually enabled or disabled.

  In the trunk drop down, there are additional options available:

  • Not Allowed: When this entry is selected, the PBX will stop processing the dial plan and reject the call. Starting with version 68, if you enter a text in the replacement field, the system will use that text as annoucement to the user why the call was rejected (in the audio language of the tenant).
  • Call Extension: This will send a call not to trunk, but to an extension. The extension must be in the replacement field. This option can be useful when using PSTN gateways behind NAT that must register with the PBX as an extension in order to be accessible through the firewall.
  • Try Loopback: When there are multiple domains, the PBX may try to send the call into another domain through a "loopback trunk". The replacement must contain a telephone number (starting with +) or a SIP URL with a username and a domain that exists on the system (since version 5.4). If that account can be located on the local system, the call will be sent there. This makes it easy to use multiple domains on the same system without an external SIP proxy.

  Pattern Matching

  Simplified pattern matching.

  There are two ways to match the dialed number, simplified and extended regular expression. Before feeding a number into the pattern matching the system represents the number in the local domain country code format. For example, in the US with the country code 1 a number 1617123123 would be reformatted to 6171231234 and +493055578992 would be formatted into 011493055578992 (unless a different presentation was selected). This makes it easy to match patterns, no matter how the user has dialed the number.

  Simplified expressions are based on the following rules:

  • Patterns can be separated by the | symbol. In this case the pattern matches if one of the pattern matches. For example 911|411 would match either number 911 or 411.
  • A literal matches if the input is equal to the parameter. For example the pattern 911 matches only if the dialed number was 911.
  • The symbol x in a pattern matches any character. For example, the pattern x11 matches 911, 411, and any other 3-digit number that ends in 11.
  • The symbol * matches an arbitrary long number of digits. When using this pattern, the replacement will use only the digits matched by the symbol.
  • Symbols in angle brackets match the characters in the bracket. For example pattern [0-7] matches characters 0 to 7, [134] matches characters 1, 3 or 4.
  • Area codes can be matches by using a area codes pattern. For example [usa] matches the domestic USA area codes (more below). This makes sure that the rule applies only to a list of codes that are hard to express in a pattern. For example the pattern [usa]xxxxxxx would match only domestic USA destinations.

  Currently the following area codes are defined:

  • [usa] domestic USA (3 digits)
  • [can] Canada area codes (3 digits)
  • [ala] Alaska area codes (3 digits, currently 907)
  • [haw] Hawaii area codes (3 digits, currently 808)
  • [fre] Toll Free USA area codes (3 digits, currently 800 888 877 866 855 844 833)
  • [pre] Premium USA (3 digits, currently 900)
  • [pue] Puerto Rico area codes (3 digits, currently 787 939)
  • [dom] Dominican Republic area codes (3 digits, currently 809 829 849)
  • [vir] United States Virgin Islands area codes (3 digits, currently 340)
  • [eur] European Union country codes

  For simplified patterns, the replacement contains literals and the * symbol to insert the match. For example, 789* would put 789 in front the number that was matched. If there is no replacement pattern, it will automatically insert the matched pattern.

  Extended Pattern Matching

  Extended pattern matching is more powerful, but it comes at the price of higher complexity. Simple and extended patterns can be mixed in the dial plan per dial plan entry.

  The input to extended patterns is in the form number@domain. The patterns are fed into the extended regular pattern matching. The matches can be referenced with the \1, \2 and so on. \r references the domain name.

  Dial Plan Samples

  Basic Dial Plan

  In many cases, you may just want to route all numbers to an outside trunk. This can be done easily by using the pattern * You don’t need to fill anything into the replacement field, the system will automatically use * as replacement.

  

  Dial Plan with Prefix in Front of the Number

  If you use a pattern like 1* in the replacement field, the system will automatically put a 1 in front of the match that it found in the pattern field. In the case when you use the pattern 617* and dial the number 6171234567 , the system will automatically convert that into a 16171234567.

  Emergency Numbers

  In the past it was common to use a prefix to tell the system that the call is an outbound call, for example a 9. While this is confusing for most callers, it is dangerous for users in case of emergency when there is no time to figure out how to make an outbound call. In some juristictions including the United States is it now the law that callers must be able to dial emergency numbers without a prefix.

  A North American Dial Plan

  If you use the Vodia PBX telephone system in the fixed-length dial plan of North America, you may use a dial plan similar to the one that follows:

  

  [911|411] The first pattern matches the emergency number and the service number explicitly. It is a good idea to have an entry for these important numbers so that they don’t accidentally get routed to the wrong gateway or are rejected. If available, these calls can be sent to a seperate trunk which might point to a local gateway.
  011* The second pattern matches all international numbers and sends them to a special trunk, which is intended to save costs for international calls.
  [usa]xxxxxxx The third pattern deals with all domestic calls. The fixed-length pattern was used so that the system can actually tell when this number is complete. The replacement puts a 1 in front of the number, putting it into a 11-digit representation.

  Sending Star Codes on a Trunk

  Extended regular expressions can be used when sending star codes on a trunk.

  

  When passing a star (*) to the carrier, a backslash is required before the star (e.g., \*67); otherwise, the system will treat the star and the numbers that follow as a star code. In the sample shown above, a backslash has been placed before the Block Caller-ID (*67 ) star code, so when the trunk is used, the system will replace the \*67 with a *67 , causing the caller-ID to be blocked. (The d in the latter part of the string tells the system to replace it with the domain. If an r had been used, the system would replace it with the registrar.)

  Important: When using star code patterns in dial plans, you must remove the code from the standard star codes and/or any customized codes. If you’d rather not use star codes in the pattern field but still want to have the dial plan activate a star code, you can use a workaround. You can create a “stand-in” star code that will be used to correlate with the desired star code. In the example shown below, users will need to enter 99 before entering the actual phone number. The system will replace it with the *67 star code. This method allows you to retain the *67 star code as part of the default list of star codes; however, users will need to be informed ahead of time so they that will know to enter 99 before entering the phone number.

  

  Extended regular expressions

  If the standard simple dialplan processing does not fit the needs, the system can use extended regular expressions instead. A typical dial plan would include the string ([0-9]*)@.* as pattern and sip:\1@\r;user=phone as the replacement. The pattern string has one group [0-9]* (which is referred in the replacement string as \1). That means that if the pattern is matched against the value 2121234567@test.com, it will store 2121234567 in the first group and the result will be sip:2121234567@test.com;user=phone (the user=phone indicates to the recipient that the number is a telephone number).

  The following dial plan illustrates the use of the caret. It’s being used to force the system to begin matching from the start of user input.

  

  Assigning dial plans

  You can create several dial plans and give them different names.

  You can select any of these dial plans as a default dial plan for a domain. The setting is Default dial plan in the domain general settings .

  When the dial plan for accounts, like extensions (see setting Dial plan), or auto attendants etc., is set as Domain Default, which is the default setting, then the above domain default dial plan will be used for that account.

  However, any other dial plan (from the list you created) can be set for each account, like extensions, or auto attendants etc.

  Text mode

  Complex dial plans can be hard to edit. In order to make this easier, there is a text mode in the dial plan web page. This mode converts the dial plan into a CSV text file that can be edited and copied e.g. into other tenants.

General Settings

Name: This setting allows you to name the trunk (the name and type of trunk can be changed at any time). The name can generally consist of any text, including spaces and special characters.

Type: There are several types of trunks available. The most common type is the registration trunk where the system behaves similar to a VoIP softphone that registers to a SIP service provider.

Direction: Traffic can be limited to either inbound traffic or outbound traffic.

• Inbound and outbound: Typical setting.
• Inbound only: Limiting traffic to inbound traffic prevents users from using the trunk for unauthorized calls.
• Outbound only: Enable this setting if you are using a trunk for outbound traffic only. It makes it easier for the system, as the trunk will not try to match inbound traffic to this trunk.

State: This setting enables you to temporarily disable a trunk rather than delete it from the system altogether. When a trunk is disabled, it cannot be used for routing, registration, etc.

Global: This setting is used only in multi-tenant environments. When it is enabled, calls that come in on this trunk are permitted to jump into other tenants if a match is found on a global alias name. (If the direction is not only inbound, then other domains may use this trunk in their dial plans for outbound calls.) When this setting is disabled, the system does not search for global alias names.

Is Secure: This setting is used to indicate that outbound calls on the trunk can be treated as secure calls. For example, when the trunk goes to a local PSTN gateway, you might decide to treat this call as a secure call. Incoming calls with the SIPS scheme will ask the Vodia PBX system to ensure that the call be kept secure end-to-end.

Inter-Office Trunk

Trunk may terminate calls for remote systems: This flag enables trunk-to-trunk calls where calls that come in on this trunk can be routed to another trunk. This is used for example when using remote offices for local termination.

Dial plan for outbound calls: When the above flag is enabled, the system needs a dial plan to determine how to terminate the call on another trunk.

Registrations

Display Name: The display name is used on SIP level when registering the trunk in the REGISTER request and (unless there is another source, like CNAM) for outbound calls.

Account: Like the display name, this setting is used on SIP level to form the From header in SIP packets unless overwritten for example in outbound calls. If is used for the user part in the URI.

Domain: Like the display name, this setting is used on SIP level to form the From header in SIP packets unless overwritten for example in outbound calls. If is used for the host part in the URI.

Username: The user name and password are used for authentication purposes. In some cases this is identical like the Account, however in some other cases it is different.

Password: The password needs to match the password from the trunk provider.

Proxy Address: This setting defines where requests that are made of this trunk will be sent. When this field is set, requests will be sent to the specified address. The outbound proxy field follows the definitions of RFC 3263 ("Locating SIP Servers"). You may use the fully qualified domain name (FQDN) for a SIP server. If you add a colon with the port number after the FQDN DNS, a resolution will be used; otherwise, the system will try DNS NAPTR and DNS SRV first. You can force the SIP connection to use TCP by supplying an outbound proxy.

Proposed duration: When registering, the PBX will propose a duration for the registration. Some SIP trunk providers look at this proposal, however the decision how long the registration is valid is made in the reponse to the registration request by the trunk provider.

Send re-registrations at this percent of expiry time: The system usually does not wait until the registration has expired. Instead it sends the request some time before the expiration. This setting controls when that is happening.

Minimum expiry time: Depending on the re-registrations percentage, the time for a re-registration can get very short. This settings limits the minimum value.

Keep-alive time: When the PBX is behind a firewall with NAT, some service providers are not able to detect this situation and come up with a registration time that will keep the NAT binding alive. In such cases this settings can override the time and explicitly set the registration time after which the system will send a re-registration. In a NAT environment, a typical value could be 30 seconds.

Proxy-Require: This setting tells the system what to put into the Proxy-Require header in the registration request. Some service providers need a text in this header to function properly.

Other headers in registration request: This setting is a generic way to add headers to the registration request. Multiple headers must be seperated by newline in the input field and the format matches the format in the SIP request (header: value). A typical example would be setting the User-Agent-header.

Send email on status change: The stability of a SIP registration is important for the overall experience of the system. In order to alert the administrator of the system, this setting can be used to trigger an email that is sent when there is a change of the SIP trunk. The system can also send registration metrics that contain more information like the response time of the SIP registrar.

DNS lookup for registration refresh: By default, the system will re-register the trunk to the same address like the last successful registration address. This avoids load on the DNS server and removes a point of failure (DNS server). However there are situations when the system should query the DNS whenever the DNS record has expired. This can be enabled with this setting.

Explicitly list addresses for inbound traffic: This setting allows you to explicitly state trusted inbound proxy addresses and enables you to identify the origin of the request. By default the system determines the list of addresses through a lookup of the proxy address (see above). The addresses can be either IP addresses (with an optional port number behind) or a DNS A or AAAA address.

Preferred SIP port for trunk: In some situations, it is neccessary to control what SIP port the trunk is using (this assumes that the trunk is using UDP). For example, in multi tenant environment this makes it possible to use SIP trunk providers that use the IP address and port to identify the tenant. The port must be in the list of SIP ports that the system opens.

Routing and Redirection

The routing of inbound calls is described in a seperate article Inbound Routing of Trunks.

Source for caller-ID information: This setting controls where the system is looking for the caller-ID for an incoming call. If the header is empty, it will fall back to the From-header.

Failover Behavior: When the trunk receives an error code, it might send the call back to the dial plan and continue the matching process. The system continues the dial plan with the next highest priority, ignoring entries with lower or same priority. This is useful when the trunk is just a "trial" to place the call. One example is when several PSTN gateways are available for terminating the call and one gateway does not accept any more calls. Another example is when you first try to route the call via a peer-to-peer call using ENUM or other location methods and only if such resolution does not result in a connection fallback to a PSTN call. The setting allows four behaviors:

• No failover: This is the default behavior. In this case, the caller will receive the error code as a result of the attempted call.
• On 5xx and 408 error codes: This will trigger failover only when a 5xx or 408 class error code is received. PSTN gateways typically return 5xx class error codes when all channels are in use; however, this mode will allow you to switch to the next PSTN gateway when a line is busy and will not trigger the failover.
• On all error codes: In this case, all error codes will trigger the failover process. Note that call redirect will also be treated as an error code, as the redirection destination can easily be abused to route calls through expensive routes.
• Always, except for busy response: Except if the remote party is busy, the system will try alternate routes.y/li>

If you are using failover, you have the option to specify a request timeout value for the trunk. By default, this is 32 seconds (the SIP default request timeout). However, in many cases, it makes sense to specify a shorter value. After the request timeout, the system will internally generate a "408 Request Timeout" response code and process it according to the failover rules.

Request timeout: The SIP standard defines when a request times out. However in some situations those timeouts are too long, e.g. waiting 32 seconds for the failover to the next trunk. This settings will trigger the timeout event at the specified time.

Accept Redirect: Use this setting if your trunk should respect redirect codes. By default, this introduces significant security risks because the system cannot determine whether these redirects introduce additional costs (redirection to expensive numbers). Therefore, enable this setting only if you are sure that your service provider or SIP gateway does not abuse this feature. This flag is especially important when you use the system with Microsoft Exchange or other Microsoft products, such as the speech server. Also enable this setting when you have a trunk that comes from another system. This will make it possible to call from one system to another.

Don't accept SIP routing changes in dialog: The SIP standard describes how the client and the server negotiate the routing of packets. However in some cases this does not work, for example when firewalls are involved or just software that is not SIP compliant. In this case the PBX will ignore routing changes and stick with the initial routing of the request.

Error message when all lines are busy: This setting determines what the system will respond when all CO-lines are in use.

Redirect destination when all lines are busy: The system can propose a redirection target for the case that all lines are busy. This is used for outbound calls from extensions.

Teams: This flag is used to tell the system that the trunk is used for the connection with Microsoft Teams.

Number/Call Identification

Prefix: This setting is used as a representation of the system when making outbound calls. If an ANI has not been set, the system puts the prefix (if there is one) in front of the primary account number and uses that as the ANI before it leaves the trunk. This is very useful in cases when a trunk deals with a range of numbers (typically outside of the NANPA area, e.g., Europe). The "extension" number is put after the prefix.

Trunk ANI: You can configure each trunk with an ANI (Automatic Number Identification). ANI is a service that tells the recipient of a telephone call the telephone number of the person making the call. In most cases, the ANI is used in the From field in the SIP packets or the caller ID.

Regular ANI rule: The rule set for determining the ANI for regular calls. For more information, see Trunk ANI.

Emergency ANI rule: The rule set for determining the ANI for emergency calls. For more information, see Trunk ANI.

SIP Caller-ID Presentation: This setting tells the system how to present the caller-ID on the trunk. The following values are available for configuring this setting:

• RFC3325 (P-Asserted-Identity): RFC 3325 describes a way to represent these two numbers. In most cases, it makes sense to use the P-Asserted-Identity header. In this case, the From header in the INVITE represents the display number, while the P-Asserted-Identity header has the network number. A similar representation can be done with the P-Preferred-Identity header. The system changes only the name of the header from P-Asserted-Identity to P-Preferred-Identity. The rest remains the same, as in the first method.
• No Indication: No Indication: This method simply discards the display number and uses only the network number in the From header. This method is a fallback when the provider cannot handle any other method. The disadvantage here is clearly that any redirection information gets lost.
• Remote-Party-ID: Remote-Party-ID: This method is described in a draft that expired years ago; however, there is still a lot of equipment outside that is supporting this method. In this case, the From header in the INVITE represents the network number, while the P-Asserted-Identity header is the display number.
• RFC3325, but don't hide: RFC 3325, but don’t hide: This method should not be used. It has been used in environments where the fields have gotten mixed up, and it is creating even more confusion.
• Custom Headers headers are discussed in detail on the Custom Trunk Headers page.

Rewrite global numbers: When you are using a trunk, you may have to represent the telephone number in a specific format. For example, in the NANPA area, you might want to use 10 or 11 digits to represent a national number. When Use + symbol at the beginning is selected, the system will automatically use 11 digits.

Country code: Sometimes the trunk "lives" in a different country than the tenant country. If set, this setting overrides the country code from the tenant.

Area code: Same like the country code, but the for area code.

P-Charging Vector: The "ICID" (IMS Charging Identifier) setting is used in the IMS(IP Multimedia Subsystem) environment. The setting is sent in the P-Charging-Vector header of the SIP packets on this trunk and it is essentially a token that identifies the trunk. If your provider uses this header, you should put it into this field.

Diversion header: This setting controls whether the system should send the Diversion-header in outgoing calls. If the header is set to {rfc} the RFC defined default is used. Otherwise the setting is subject to variable replacement like in the Custom Trunk Headers.

Avoid RFC4122 (UUID): UUID are a way to identify the SIP trunk in a unique way. This helps to avoid double registration, e.g. after a restart of the system. Like with many useful features, many SIP service providers don't support the feature and reject the request. In such cases, the UUID must be suppressed.

Attach PIDF-LO for emergency calls: PIDF-LO objects can be used to indicate the location of the caller. The system can use this to indicate the location for emergency calls in a multipart-MIME attachment for outbound calls.

Show History-Info: The History-Info-header is a powerful way to indicate how the call was redirected inside the system. Some service providers are able to use this header to provide additional information to the caller.

Indicate RTCP-XR support: RTCP-XR is a powerful way to measure and report the call quality for example for MOS, however it requires that both sides of the call support the standard. Although negotiated automatically, many SIP servers get confused when RTCP-XR support is proposed and reject calls. Because of this, the support indication can be suppressed.

Encrypt media: By defaut the system will propose that RTP is encrypted (SRTP) when a secure transport layer such as TLS is used for the call setup. However some SIP servers get confused with this. In such cases this setting can be used to suppress encrypted audio.

Generate unique extension identifier: This setting allows you to use the extension’s EPID instead of the ANI if the extension’s EPID has already been set. If extension’s EPID is not set, "generate" a new EPID and use it (and set it for extension for the future use too).

Interpret SIP URI always as telephone number: When a call comes into the system, the system needs to know how to interpret the number. In SIP, the URI contains a domain name; however, in most cases, the domain name should be ignored when interpreting the URI coming from this trunk (because the user portion is just a telephone number). Usually, this is indicated by the parameter "user=phone," but not all service providers set this flag. By turning on the "Interpret SIP URI always as telephone number," you make the system believe that this flag was set on the trunk call.

Caller ID update on trunk: This setting is used to update the caller ID information on the trunk. For example, if you want to update the caller ID that is sent to a remote party, when a call is transferred, you can use this setting. This is dependent on whether the trunk provider supports that feature.

Lookup SPAM provider if available: When a SPAM provider was set up on the system, this setting controls if that provider i used to look the caller up when a call comes in through the trunk.

Contact header value: Some SIP servers get confused with the standard SIP contact header value. This setting can be used to override the value of the contact header.

Customer specific header: Like with the registration, additional headers can be specified for the trunk for outbound calls. The format is the same like with the registration. In contract to the registration header, this setting is subject to variable replacement, for example caller-ID information can be inserted in to the requests (e.g. X-Other-From: {from}).

Media/Audio

Codec preference: Use this setting to specify particular codec preferences for the trunk. By default, the system uses the codec preference of the system, which has the advantage of allowing the system to negotiate codecs in such a way that transcoding can be avoided. (When default mode is used, no codecs will be displayed on the left side of the codec selection box.) If it is necessary to enforce specific codecs on a trunk, choose a codec from the list of available codecs and click theAdd button. Repeat as needed. Be sure to position the codec with the highest preference first. To delete a codec, click the Remove button. To move a codec up or down in the preference list, use the Up and Down buttons.

Lock codec during conversation: Locking the codec prevents someone from switching codecs in mid-conversation and ensures that the user agent continues to send and receive media properly. (Some user agents are not capable of changing the media once the conversation begins. ) When this setting is used, the Vodia PBX transcodes the stream so that changes made to the codec by the other side go unnoticed by the user agent.

Number of codecs in SDP answer: Some trunks cannot handle multiple codecs in a SDP answer. In this case it makes sense to limit the number of codecs to 1.

Generate PCAP traces for calls. When this flag is turned on, the PBX is generating PCAP files for the call which can be retrieved from the system call log or from the file system.

Strict RTP Routing: This setting is needed because the IETF allows "RTP traffic send ports" to be different from "RTP receiving ports," creating an extremely NAT-unfriendly situation. While most implementations today use the same port number for sending and receiving RTP, some gateways still insist on strict IETF compatibility. In such cases, this setting should be enabled. We recommend you keep this flag disabled ("No") unless asymmetrical ports are required.

Port range start, end: This setting overrides the setting for the RTP port range on system level. Some SIP trunks need the RTP ports to be in a specific range; if that range is very short it makes sense to set it for the trunk and not globally for the whole system.

Trunk requires out-of-band DTMF tones: When this setting is enabled, the trunk will use RFC 2833.

Requires busy tone detection: When an analog PSTN gateway (e.g., FXO) is used, hangup detection can be an issue. In FXO, the hangup signal might be just a tone that needs to be detected. Unfortunately, no international standard exists for the disconnect tone. Incoming international calls might give you a disconnect tone that the system has to recognize. Of course, if the PSTN gateway is capable of detecting this, the task should be left to the PSTN gateway. However, as a fallback, you may also configure the system to perform the hangup detection. The disadvantage of doing this is that it costs additional CPU resources, and it might randomly disconnect calls; for example, if the other party plays back a tone that sounds like a busy tone, the call may be disconnected. The best way to avoid these kinds of problems is to use a digital line, e.g., a SIP trunk.

Use T.38 for FAX: By default the trunk will attempt to use T.38 for FAX transmission. However in some cases, modem encoding must be used.

PRACK support: Not all SIP trunks support PRACK, which make the signaling of early media more reliable. This flag can be used to suppress PRACK.

Ringback on incoming calls: This feature was introduced to deal with network operators who were unable to deal with early media. Although using the 180 Message simplifies the signaling in forking calls scenarios, it creates additional delay when the called party picks the handset up and the first samples on the conversion may not be transported. We strongly recommend leaving the flag set to Media (which is default) and asking the operator to fix the problems with early media.

Local ringback generation for outbound calls: This setting is used in cases where the service provider sends back a 183 with SDP, but doesn't provide the ringback tone. With this setting, the PBX can force a local ringback. For providers that support P-Early-Media it is suggested to turn this feature on, so that the PBX can automatically determine wether to play local or remote ringback.

Play file for local ringback when needed: There are conflicting cases when the PBX should actually play local ringback, but the trunk is still providing it. There are only few cases when this flag needs to be turned off.

Inbound Routing of Trunks

6173998147

ext-ani 6173998147

!disa:40*:dom-ani ext-ani 6173998147

acd-ani ext-ani trunk-prefix dom-ani trunk-ani from

ext-epid ext-eani dom-eani

Canada,+1204,0.1
Canada,+1250,0.1
Canada,+1289,0.1
Canada,+1306,0.1
Canada,+1403,0.1
Canada,+1416,0.1
Canada,+1418,0.1
Canada,+1450,0.1
Canada,+1506,0.1
Canada,+1514,0.1
Canada,+1519,0.1
Canada,+1604,0.1
Canada,+1613,0.1
Canada,+1647,0.1
Canada,+1705,0.1
Canada,+1709,0.1
Canada,+1778,0.1
Canada,+1780,0.1
Canada,+1807,0.1
Canada,+1819,0.1
Canada,+1867,0.1
Canada,+1902,0.1
Canada,+1905,0.1
USA/1000,+1,0.05
Greece,+30,0.10
Germany,+49,0.15
Cuba,+527,0.45,55,1.00
Rest of World,*,0.25

Call forwarding allows users to forward their calls to another number. Multiple methods are available for doing this.

Hot Desking (*70)

Hot desking can minimize the need for office space and is often used as a cost-reduction method. It allows multiple users to use the same desk and phone, although at different times. Hot desking can be used on a temporary or permanent basis. In either case, once the user activates the hot desking feature, all calls made to the user’s extension (or as part of a hunt group or agent group) will be routed to the new extension or destination. Outbound calls from the hot desking phone will reflect the user’s caller-ID.

It is not expected that other significant inbound traffic is routed to the originally registered extension. This fact suggests that offices use “virtual” and “real” extension numbers:

• Real extension numbers are used for employees with a fixed location (e.g., switch board, management).
• Virtual extensions are not registered. They are used simply for routing calls to a specific user.

To log into a hot desk:

1. Go to the phone where you will be hot desking.
2. Dial *70 and press the confirm button on the telephone keypad.
3. Enter your extension number when prompted.
4. Enter your voicemail PIN code for the extension number when prompted.

The system will acknowledge that the Hot Desking feature is in service.

To log out of a hot desk:

To log out of a hot desk, users must dial *70 and their extension number from either the location that is currently registered as a hot desk or from a phone that has been configured to their extension. The system will acknowledge that hot desking is no longer in service.

Call Forward All Calls (*71/*72)

The Call Forward All (*71) feature allows users to redirect all calls to another number (e.g., to a colleague who is answering the user’s phone).

1. Dial *71 and press the confirm button on the telephone keypad.
2. Enter the call forwarding number once the announcement is complete.
3. Press the # key. The system will repeat the number and then hang up. If you want to change the forwarding number, just call *71 again.
4. To deactivate, press 72, then press the * key when prompted. The announcement will indicate that call forwarding has been turned off.

Users do not need physical access to their phone to modify this setting. This setting can also be activated from the Redirection settings of the web interface.

Call Forward All Calls (*71/*72)

The Call Forward All (*71) feature allows users to redirect all calls to another number (e.g., to a colleague who is answering the user’s phone).

1. Dial *71 and press the confirm button on the telephone keypad.
2. Enter the call forwarding number once the announcement is complete.
3. Press the # key. The system will repeat the number and then hang up. If you want to change the forwarding number, just call *71 again.
4. To deactivate, press *72, then press the * key when prompted. The announcement will indicate that call forwarding has been turned off.

Users do not need physical access to their phone to modify this setting. This setting can also be activated from the Redirection settings of the web interface.

Call Forward on Busy (*73/*74)

The Call Forward on Busy (*73) feature allows users to forward calls to another number if their extension is busy. This feature is typically used for redirecting calls to a team member or colleague to ensure that all incoming calls are answered. To activate:

1. Dial *73 and press the confirm button on the telephone keypad.
2. Enter the call forwarding number once the announcement is complete.
3. Press the pound key #. The system will repeat the number and then hang up. If you want to change the forwarding number, just call *73 again.
4. To deactivate, dial *74. The announcement will indicate that call forwarding has been turned off.

Users do not need physical access to their phone to modify this setting. This setting can also be activated from the Redirection settings of the web interface.

Call Forward on No Answer (*75/*76)

The Call Forward on No Answer feature allows users to redirect calls when the extension has not picked up after a certain time. This feature is typically used to redirect a user’s incoming calls to an assistant or secondary phone. To activate:

1. Dial *75 and press the confirm button on the telephone keypad.
2. Enter the call forwarding number once the announcement is complete.
3. Press the pound key #. The system will repeat the number and hang up. If users want to change the forwarding number, they must dial *75 again.
4. To disable this feature, dial *76. The announcement will indicate that call forwarding has been turned off.

The length of time the system will wait before forwarding the call is determined by the also be activated from Call forward on no answer timeout setting of the web interface. Keep in mind that the voicemail timeout (see Time until user’s mailbox picks up ) works in tandem with the no-answer timeout. If the voicemail timeout kicks in before the no-answer timeout, the call will go to voicemail. Both of these settings can be set at the domain, but users can override these settings.

Do Not Disturb (*78)

Do Not Disturb or DND allows users to silence their phone for all incoming calls, even calls made to a hunt group or agent group. When DND is activated, all calls will stop, including those redirected to the cell phone. However, the system will still call the user’s cell phone though if a message has been received while DND is activated and the user has activated Call cell phone when new message arrives.

1. To activate DND, dial *78 and press the confirm button on the telephone keypad.
2. Dial *79 to deactivate.

Set Night Mode for Domain Accounts (*80)

This feature allows users to forward domain accounts (e.g., auto attendant, hunt group, etc.) to their own extension. Before this feature can be used, permission for each account must be activated by the administrator. Permission is set from the extension's Permissions tab.

• To forward all auto attendants, hunt groups, and agent groups of the domain, *80 must be used without an argument.
• To forward only a specific account, the account must be specified after the star code.
• To disable this feature, enter a star in place of the call forward number.

  Cellphone Codes

  Feature Codes

  The feature codes used in this section are related to cell phone use (*51, *52, or *00) but they cannot be executed from the cell phone.

  Retrieve Call From Cell Phone (*51)

  This feature allows users to pick up an in-process cell phone call on their extension phone. This feature can be used only for calls that have been made from the cell phone (through the Personal Virtual Assistant) or which have been redirected to the user's cell phone. Calls that have been made directly to the user's cell phone (without involving the system) cannot be picked up on the user's extension phone.

  Scenario : You're on your way to the office and you need to call a client, so you call into the system and place an outbound call. You speak with the client on your way to work, and once you arrive at the office, you decide to pick the call up from your office phone.

  To transfer a cell phone call to your extension:

  1. From your cell phone, call into the main number of the PBX (your cell phone must be associated with your account).
  2. From the Personal Virtual Assistant menu, press 1 for “outbound call.”
  3. Enter a destination number, then press the # key.
  4. When you're ready to pick the call up from your office phone, dial *51 from the office phone.

  Once the call has been connected, your cell phone will disconnect and will no longer accumulate minutes.

  Move Current Call to Cell Phone (*52)

  This feature allows users to move a current call from the extension phone to the cell phone. (Note : The user's cell phone must be configured to the extension before this feature can be used.)

  Scenario : You're on your extension in the middle of a conversation and need to leave the office. Instead of ending the conversation, you put the caller on hold and transfer the call to your cell phone.

  To transfer an extension call to your cell phone:

  1. Place the extension call on hold.
  2. Dial *52 from your office phone.
  3. Answer the call on your cell phone.

  Call Cell Phone of the Extension (*00)

  This feature allows users to call someone else's cell phone without knowing that person's cell phone number. (Note : The individual's cell phone number must be configured as part of the extension that is being called.)

  Scenario : You notice that your boss has just left the building when an important call comes in. You have no idea what his cell phone number is, but you must speak with him. You know what his extension number is, so you use the *00star code followed by his extension number. Because the system sees the extension and the cell phone number as being the same, the system is able to ring his cell phone.

  To call someone's cell phone without knowing the number:

  1. Dial *00 followed by the person's extension number.
  2. Press the start button from the telephone keypad.

  The system will connect you to the individual's cell phone.

  Making Calls Codes

  Redial a Number (*66)

  Redial (*66) allows you to redial the last number that was called from your phone.

  Call Return (*69)

  Call Return (*69) will dial the number of the call that was received last. The number will be available until a call has been established to that number—not necessarily connected, which can help prevent users from calling back the same person twice. (If you execute *69and discontinue the call during its ring state though, *69will not be able to access that number again unless the caller rings again.)

  Notes : If you reach an external mailbox, the system will handle the Call Return as if the call had been successfully established and will clear the number. If you reach a mailbox of another extension on the system, the Call Return number will not be cleared and you will be able to make successive attempts at reaching that extension by dialing *69. The system will store a Call Return number only for calls that contain a valid caller-ID.

  Telephones that share the same extension number will share the same redial and Call Return number.

  Intercom (*90)

  The Intercom feature mimics the old secretary-boss intercom systems and can be thought of as a one-part page with two-way audio. Intercom is used to communicate with one other person. The other party must have a separate extension number, as the Intercom feature will not work between two phones that are registered to the same extension. The Intercom feature will also not work when multiple registrations are involved with the receiving extension.

  To activate:

  1. Set the user’s Call Permission settings for intercom.
  2. Dial *90and the extension you would like to intercom.
  3. Press the confirm button on the telephone keypad.

  You can now begin conversing with the other party.

  Miscellaneous Codes

  Conference ( *53 )

  The conference code can be used to put the callers that the current extension is holding into a conference room. If the code is used without any digits behind it, the system will search for the first conference room that allowed non-scheduled conferences and puts all callers into that conference room. If the caller puts a number behind it (e.g. *53800), then the PBX will use the indicated conference room (e.g. 800). The conference room must be suitable for this, otherwise the system will play back an error message.

  Outbound Calling ( *55 )

  This feature is described in a separate page, see Agent Group Outbound Calling

  Play Recordings ( *56 )

  This code can be used alone or with a number behind it. When used without a number behind it, it will start playing the recorded messages for the extension from which the code was called. Otherwise it will play the recordings for the account that was provided behind the code. For example, to play back all recordings from the ACD 400, the user would dial *56400. The user needs to have the permission to play back the recordings (see Extension Call Permissions for this).

  The playback starts with the last message and then moves backward in time.

  When listing to the recordings, the user can use the following keys to change the playback:

  • # can be used to move to the previous recording.
  • * be used to move to the next recording.
  • 7can be used to move ten recording forward.
  • 8 can be used to move start the current recording from the beginning.
  • 9 can be used to move ten recording backward.
  • 1 can be used to reverse the playback. Pushing the button several times increases the speed.
  • 2can be used to stop and resume the playback.
  • 3 can be used to fast forward the playback.

  Show Account Balance ( *61 )

  This feature allows users to hear the balance of their pre-paid calling card account. To get the balance, users simply enter *61 (followed by the confirm button) from the extension the pre-paid card is configured to.

  Wakeup Call ( *62 )

  The Wakeup Call (*62) feature allows users to receive automatic wakeup calls from the system. Though this feature is used primarily in hospitality environments, it can also be used on an individual basis.

  1. Dial *62.
  2. Press the confirm button.
  3. Enter the wakeup time in 24-hour format (HHMM); for example, 0715 (for 7:15 a.m.) and 1915 (for 7:15 p.m.).

  The system will confirm the wakeup call. To cancel a wakeup call after it has already been confirmed, users must press the star (*) key.

  Note : Wakeup calls cannot be transferred to other registered devices (i.e., the wakeup call will not ring the user's cell phone).

  Request Call Details ( *63 )

  The Request Call Details (*63) feature allows users to request the details of a call (caller-ID, call duration, and time of call). The information will be sent to the user's email address. This feature is handy and prevents the user from asking for contact information from the caller.

  To activate the Request Call Details feature:

  1. Dial *63and press the confirm button on the telephone keypad.
  2. Retrieve the message in your email.

  Call forward all calls via Cellphone (Cellphone tied to the user's Extension)

  This helps to call into the system and change the calls to be forwarded settings, just by calling into the system via VPA (Virtual Private Assistant).

  To activate "Call forward all" star code feature:

  1. Make sure VPA feature is on the system.
  2. Call from the user's cell phone (which is part of his extension on the PBX) into the system.
  3. Press 3, to go to the voicemail.
  4. Enter the Mailbox PIN if prompted (or you can skip this step).
  5. Press **71<cell phone number to forward calls to># (in one continuous string)
  6. And same for turning the feature off. Just change step 5 to include *72 instead.

  Clean up an Extension ( *84 )

  This feature allows users to quickly clean up an extension. This makes it ideal in hospitality environments, as it can be used to clear an extension of its messages after guests have checked out and before new ones arrive. It can also be used to clean up extensions of employees who have left the company. Users cannot activate this feature unless permission has been given by the domain administrator.

  Add to White List ( *91)

  This feature allows users to add a contact to the white list of their personal address book. White list contacts receive preferred treatment. They are not required to say their name before speaking with the user, and they can receive a callback when the user's extension becomes available.

  1. Dial *91.
  2. Press the confirm button.

  If the number does not already exist in the user's address book, the system will automatically create an entry for the number. Users can edit the address book entry from the web interface.

  Add to Black List ( *92 )

  This feature helps users avoid unwanted contacts (e.g., unwanted callers, fax spammers, etc.). Once a caller is on the black list, the user's Incoming anonymous call setting will determine how that call is treated. If the caller should be rejected, then the system will reject that call. Otherwise, the system will carry out whatever instruction the user has configured.

  If the number does not already exist in the user's address book, the system will automatically create an entry for this number. Users can edit the address book entry from the web interface. Black list contacts are not allowed to camp on an extension.

  Prior to version 5.6, you need to reject the call or wait until the caller hung up to be able to add the caller. Since version 5.6, the system will take the last number that was calling the extension, unless it was more than 5 minutes ago.

  Call Record ( *93/*94 )

  Call Record allows users to record a conversation between themselves and another party. This feature is handy when it is critical that a user remember the contents of a particular call. Users should check state and federal laws before recording calls. To activate the record feature, dial *93on the telephone keypad. To deactivate, dial *94. The recording can be retrieved from the user's mailbox from the web interface.

  If automatic call recording is in progress, the codes are used to pause and resume the recording. This is useful when sensitive information like credit card information should not be recorded in the conversation. In this case the agent would first enter the deactivate code *94, then listen to the credit card information and when done resume recording with *93.

  Set ANI (*59)

  This feature allows the extensions to dial the star code and change their outbound ANI on the go. Steps are:

  1. Make sure that the star code for Set ANI is set, e.g. to *59. This step may be neccessary when upgrading from older versions.
  2. Add ANIs to the desired account.
  3. Go to Domain → Setting → General Settings → Extensions may use star code to change ANI field and choose from either of those drop down menus:
     1. Only own ANI: User / Extension can enter only it's own DID number (as is) as it's Outbound ANI number.
     2. Any ANI in this Domain: User / Extension can enter any DID number as their own ANI that they can find inside the domain that they're a part of.
     3. Any NANPA ANI: User / Extension can enter any DID number that belong to the NANPA number format on whole of the PBX.
     4. Enter any ANI: User / Extension can enter any DID number that resides on the PBX as their own ANI.
  4. Dial e.g. *591234567899 (use your own ANI) and hit send and the audio prompt will confirm that ANI has been set by repeating the number to you, if successful.
  5. If it doesn't go through, it will prompt the user with "This number could not be found".
  6. Make a call out. This will only work, if your trunk allows random ANIs to go out, or else it wont.

  Echo Test (*56)

  When calling the echo test, the system will send the received audio back to the calling device. This is a simple way to test if the extension has two-way audio. In addition to that the system will read out DTMF that is has received, which helps to make sure that the device works properly with negotiating the DTMF codec.

  Transferring Calls Codes

  Calls can be transferred using the transfer button on the phone or star codes. The system supports two types of call transfers: blind transfer and attended transfer.

  Blind Transfer

  When the user activates a blind transfer, the call is transferred with no knowledge as to whether the receiving party will be available to receive the call. A blind transfer can be used for transferring calls to voicemail or the cell phone.

  1. Press the transfer button on your phone, or press Hold and dial *77.
  2. Enter the extension number (or phone number) of the recipient.

  A blind transfer offers no confirmation of call connectivity. If the number is busy or does not exist or no one picks up, the user will not be notified.

  Attended Transfer
  When using attended transfer, the user must speak with the party who is receiving the call to ensure that the call can be received.

  1. Put the caller on hold (using the phone’s hold button). The LCD will display the call that is on hold.
  2. Enter the destination number,then press the confirm button.
  3. Once the party on the receiving end is prepared to take the call, press the transfer button twice (the first Transfer connects the call; the second Transfer clears the music so that the parties can speak. (If the phone does not have a Transfer button, *77can be used to transfer the call.)

  Users do not need to press the hold button to free the call. If the person is not available to take the call, the user can press the Hold button to reclaim the call and discuss the available options with the caller.

  Transfer a Call Directly to Someone’s Voicemail

  The systems administrator must enable a direct dial prefix before users can "blindly" transfer a call to another user’s voice mail. The default for direct dial is "8".

  1. Put the call on hold using the hold button on your telephone keypad.
  2. Press the Transfer key. (If your phone does not have a transfer button, dial *77to transfer the call.)
  3. Enter the mailbox prefix number.
  4. Enter the extension number.

  The caller will be transferred to the extension’s voicemail. Because the caller will bypass the voicemail timeout, the caller will not hear any ringing. This method does not need to involve an outside caller. To call a person’s mailbox directly, the user can dial 8123.

  Call Park ( *85 ) and Call Park Retrieve ( *86 )

  The system administrator must configure specific park orbits (e.g., 800, 801, 802, and 803) on the system before these star codes can be used. Call Park allows the user to temporarily park a call so that it can be picked up by another user. The receiving party uses Call Park Retrieve (*86), followed by the park orbit number, to retrieve the call.

  Scenario : A call comes in for Diane, but you do not know where she it, so you put the customer on hold, dial *85and the park orbit number, and press the confirm button. You then page Diane, "Diane, you have a call on extension 509, (509is the park orbit on which the call was placed). Diane executes the Call Park Retrieve (*86) feature, dials 509, and retrieves the call.

  An announcement will indicate that the call has been parked. At this point, the call can be retrieved by any extension by dialing *86and the park orbit number on which the call has been parked.

  1. Put the call on hold (music will play in the background).
  2. Press *85to park the call.
  3. Enter the park orbit number.

  If the call is not picked up within a specified time (default is 1 minute), the user will receive a call from the system as a reminder of the waiting call.

  Note : To create a toggle-like effect for Call Park and Call Park Retrieve for use with buttons, use matching star codes (e.g., *85/*85or *86/*86) on the star codes page at the domain level.

  Call Pickup ( *87 )

  Call Pickup allows users to pick up a ringing phone. It can be used to pick up calls ringing into a specific extension or for ringing calls in general, including calls ringing into a hunt group or agent group.

  Note : When compared to Call Park Retrieve, Call Pickup is associated with a sense of urgency (i.e., the user must pick up the ringing phone before the call goes to voicemail.) With Call Park Retrieve, the caller is already in the system, so there is not as much of a hurry. Before users can use Call Pickup, they must first put their last call on hold so that the system does not interpret and execute the star key during calls made to external systems that also use star keys.

  Not everyone can pickup every call. Only users that have the "Permission to monitor this account" may pickup calls from the account (see General Extension Settings ).

  Directed Pickup

  1. Dial *87.
  2. Dial the extension on which the call is ringing.
  3. Press the confirm button on the telephone keypad.

  General Pickup

  Call Pickup can also be used without specifying an extension. To activate, dial *87and press the confirm button on the telephone keypad. The system searches for calls to pick up in the following order:

  • Hunt groups: For a call that is headed for a specified hunt group, the system will reroute the call to the extension that dials the pickup code. If several calls ring the hunt group, then the system will pick up only the first call that was routed to the hunt group.

  • Extensions: When the account number is an extension, the system searches for calls that go directly to the specified extension. These calls can come from an auto attendant or through direct extension dialing. Calls that ring the extension due to it being part of a hunt group or agent group will not be affected by Call Pickup.

  • Agent groups: As with the hunt group, the system picks up a call from the specified agent group. For calls in the ringing state, the system picks the first call that is in that state. Otherwise, the system picks the call that entered the queue first.

    Voicemail

    You can activate "Send Voicemails as Emails", using the star code *95

    You can deactivate Voicemail, using the star code *96

    You can go to Voicemail, using the star code *97

    You can record a new greeting, using the star code *98. To delete a recording, you can press * after the call was connected, or press 1 to discard the current recording and record it again. For the recording, there are multiple ways to start a recording:

    • *98: This starts to record the standard annoucement for the mailbox.
    • 98 n : This starts to record a specific annoucement for the mailbox. If n is 0, then the name will be recorded; if it is between 1 and 5, it will record the personal greetings 1 to 5. The number 9 will record a mailbox message, similar to a note pad.
    • *98 m : This starts to record the standard annoucement for an account, which must be an auto attendant, a IVR node, a ACD or a paging account.
    • *98 m * n : When used with an auto attendant, where m is the account number and n is the index into the annoucement which you want to record. The same is possible for th ACD, where the index n can be between 0 (the initial greeting) and 9.

    You can clear the voice message indicator, using the star code *99

    You can also listen to automatic recordings using *56 or turn on/off recording using *93/*94 star codes from the telephone keypad.

    LAN Provisioning With Single Button Press

    • Login as Admin.
    • Click on Setup Devices in LAN as shown above.

    

    Automatic Detection and Setup of VoIP Devices

    The page will show you the phones that were found in the LAN, which you can assign to an extension. Select the phones that you want to provision/setup, select a domain and an extension for each phone and click on the Setup button. The setup requires that the phones use their default username and password. Therefore it is recommended to factory reset the devices and wait till they completely come back up.

    In case the device is a PSTN gateway, select the domain in which you want to set it up and select the extension where all the calls from the PSTN device will be directed, for example, to an Auto Attendant. Also, for a PSTN gateway, choose the number of lines from the drop down next to it for the number of active lines (PSTN lines actually connected) to the gateway.

    Note : Give the devices some time before they start provisioning.

    Setup of Static Devices

    In case the device has a static IP (like VoIP gateways) and is therefore not detected, then it has to be first added to the list before setup. For that, first press the button Add Device with Static IP to reveal other settings then select the device (gateway) vendor, enter its MAC (e.g. AA:BB:CC:DD:EE:FF), its static ip, its subnet mask, its default gateway (exactly as they are set on the device) and press "Add". The device will be added to the list, which you can select and "Setup" accordingly, as described above.

    Plug and Play Settings

    Automatic provisioning of VoIP phones is a key functionality of the PBX.

    Write PnP generated files. When the system generates provisioning files, it can write them to the file system or send then to the log file. When writing them into the file system, the system will generate a folder generated and create directories for the domain and the extension. Writing them into the file system makes it easy to trace what has been sent to a specific extension, the drawback is that it requires file system access to read them and it also exposes passwords on the file system. In any case the generated files are kept in the extension tab in the domain web interface.

    Duration for opening extensions for MAC-based provisioning. When a MAC address gets opened up for provisioning, any device that knows the MAC address can pull the configuration data for that device. Because a MAC address contains only a relatively small number of bits, the window where that can happen should be kept as short as possible. Otherwise scanners can accidentally guess a valid address and pull the configuration data for the device, which usually includes the password for the SIP registration. The time should be especially short when using a wildcard for MAC address, because then there is no need to guess a MAC address and a scanner will immediately find a valid address.

    Duration for keeping provisioning open. In the process of pairing a MAC address, the PBX will keep the provisioning window open for some time. By default this time is 5 minutes. This settings allows changes of that default.

    Duration for after provisioning passwords. Like the previous setting, this settings controls the duration how long the PBX will send passwords after it has sent the first password to a device. Usually this value can be rather short, but in some cases it is necessary to increase the duration.

    Automatically list unassigned MAC addresses. By default, the PBX keeps track of MAC addresses that hit the PBX provisioning system without being assigned to an extension. This makes LAN provisioning very easy, however in hosted PBX environments it usually is not of great use, especially if the system is running for a very long time.

    Automatically open extensions after creation. When creating an extension, in many cases the next step is to plug in a phone and provision it. If this setting is turned on, the system will automatically open that extension up for provisioning for the duration set by the previous setting.

    Use domain name instead of IP address. By default, the PBX figures out what IP address was used to connect to the PBX and then use that IP address when generating provisioning files for a VoIP phone. This is problematic when the PBX should also work after a failover, when the backup systems runs on another IP address. In that case, using a DNS address that matches the domain name can solve the problem; the setting will then provision the DNS address.

    Try to provision phones securely. For some phones, it is possible to provision phones securely or without a secure connection. This setting controls if secure provisioning should be attempted.

    Show advice of charge (AOC) on phone. Some devices are able to show how much a call costs (advice of charge). When using a trunk that has rates set up for it, the PBX can update those devices about how much the call has cost according to those rates. Also, some trunk providers provide AOC as part of the service that the PBX can pass through to the connected phone.

    Label template for busy lamp fields. By default, the PBX will send patterns in the form {user} {name}to the phone when it wants to label BLF buttons. However the administrator may choose a different pattern in this setting.

    Label template for private lines. As with the setting for the BLF buttons, this settings defines how private lines are labeled by default. As with the BLF setting, the label can be overridden on button template level as well.

    Characters to be stripped from dialed numbers. Certain devices, mostly soft phones, tend to send additional characters in the dialed numbers that are used for displaying phone numbers. For example, when dialing 6173998147 a phone might actually send (617) 399 4147 instead. This setting lists the characters that will be removed from the dialed number.

    Modifying Plug and Play Files

    For the most part, the Vodia PBX takes care of provisioning of the phones with the its default configuration. But in some cases customers want to tweak some of the phone settings for various reasons. This chapter explains different mechanism the system administrator can adopt to achieve that goal.

    Many templates contain a general parameter that you can set with additional content that you would like to add to the template. If you can put you own content in those parameters, you don't have to modify the templates. This has great advantages when you upgrade the PBX software, as you will not miss those changes.

    If you really have to edit the template make sure that add a comment that will help you later find out what you have changed. This is important if an upgrade makes your changes incompatible. As this can cost a lot of time, it is important that you keep the notes in the comment field to save time figuring out what has changed.

    Types of Phones provisioning

    A. DHCP Option 66 Provisioning

    • Enter the PBX IP address in your Local Router as "http://PBX_IP_ADDRESS"
    • Turn on the DHCP 66 option on the phones.
    • Reset the phone to factory default settings, add the Mac address under the "provisioning" tab as shown above, open the phone for "Mac based provisioning" and then reboot the phone.

    B. Manual Provisioning

    • Step 1 : Add the Mac address of the phone into the Provisioning tab of the extension and hit Save.

    

    • Step 2: Under Domain> Advanced> General Settings> Provisioning Parametersas shown below, add the authentication password and PIN to login into the phone (after it gets provisioned).
    • Select the Dial plan scheme for your country as well from this section.

    

    • Step 3: Select MAC based Provisioning

    Navigate to the extension level and select "Open account for MAC based provisioning", to allow the phone to provision to that specific account.
    

    • Step 4: Navigate to the web GUI of the phone and add the URL of the PBX Server as below:

    

    • Step 5: Reboot the phone. The phone might take 2-3 reboots before it is up and registered.

    C. LAN Provisioning

    • Step 1: Navigate to Admin> Phones> LAN Devicesand select the MAC and click on Setup.

    

    • Step 2: Assign it to domain and the extension inside that domain as shown.

    

    • Step 3: The PBX should automatically add the MAC address of the phone to the extension, reboot, provision and register !

      PnP Pre-requisite Steps

      Many Vodia PBX settings are downloaded to the phone during Plug and Play, so it's best to configure these settings before actually doing Plug and Play. That way, when the phones are provisioned, they will have those settings. Some of these settings are listed below.

      Configuring the Domain Settings for Provisioning Settings

      Many domain settings are also uploaded to the extension (the phone) during plug and play, such as language settings and country codes etc.

      

      • Go to your selected domain in the Vodia PBX web interface.
      • Go to General Settings
      • Choose default IVR and web languages, a time zone, and country and area codes etc.
      • Scroll to the bottom of the page to Provisioning Parameters , and select a default dial plan scheme and enter the username and password for the phones.
      • Dial plan has to be selected here at this point.
      • The Authentication user , Authentication password and Authentication PIN settings allow administrators to provision phones and log into a phone’s web interface remotely. The password will bypass users’ passwords (the web interface passwords), so users will not be able to use their passwords; hence, this setting can be used to keep users out of the phone’s web interface and prevent them from inadvertently changing anything. The default username is admin , and the default password is <blank>.

      Configuring the Domain Button Profiles

      Not all users will need the same button profile, so you may want to set up multiple profiles and assign them to the extensions that will be using them.
      There are also several default button profiles available for the different Snom phones, which can be changed by the user accordingly, and which are provisioned to the phones according to the phone types when the button profile setting on the extension level is set to “Automatic” instead of being assigned to a specific profile.
      It is explained in more detail under Buttons

      Configuring the Extension(s)

      Many extension settings are also uploaded to the extension (the phone) during plug and play. The extension based settings can be set from the domain (while logged in as admin and selecting the extension for which changes are to be made) and also from the user portal (logged in as a user).

      

      

      • Go to your selected domain in the Vodia PBX web interface.
      • Click on the extension from the list for which you want to change settings.
      • For that extension you will see several tabs, which have related settings that you can change for PnP, e.g. General tab for some general settings, Registration tab for some registration specific settings including binding a phone to a specific MAC and Buttons tab for selecting or customizing a buttons profile or selecting “Automatic” for automatic selection of the default button profile based on the Snom phone type (for more details on buttons, go to Buttons . All these settings will be uploaded to the phone tied to this extension on provisioning.
      • You can also login as a user (User Mode) to the user portal and make changes to the settings for PnP, which will be uploaded to the phone during provisioning.
      • You can tie a phone to an extension by adding the phone’s MAC address to the extension. Under the tab Registration for the extension, enter the MAC address of the phone you want to tie to that extension in the setting Bind to MAC Address and press Save. This will tie the phone to that extension and when it is provisioned, the phone will get all the settings from this extension, including registration information, buttons, and everything else that is provisioned.
      • Three different options are available to associate the phone’s MAC address with an extension: manual, permanent assignment, and temporary assignment. These methods are detailed below.

      PnP Method Use Case Scenario How it's Used
      Manual Setting up a few phones In this mode, you can permanently set the MAC address in the extension account. This will bind the registration to a specific phone. The MAC address can be used in multiple extensions, and multiple MAC addresses can be assigned to the same extension. This allows for users to have multiple phones. (Use a space to separate the MAC addresses.)
      Temporary Assignment (?)Setting up phones at trade shows, classrooms, etc. This mode simplifies preparation when setting up at trade shows and for classroom training sessions. When this mode is used, the system will not remember which phone had been assigned to which extension. A phone might receive a new extension number every time it boots up. (As long as no other user agent is registered to a particular extension, the extension will be available for plug and play.) This mode requires that a question mark (?) be used in the Bind to MAC Address field.

      Outbound Proxy Provisioning

      The information on this page is based on versions after 5.4.2 (May 27, 2016).

      Available Transport Layers

      The PBX can automatically generate the configuration files for a range of VoIP phones. Those configuration files also tell the phones to which address they should register, and what protocol to use. The PBX supports UDP, TCP and TLS transport layers:

      Protocol	Description
      UDP	The UDP prococol must be support by all devices that claim to be SIP compliant. This transport protocol serves as the least common denominator. However for registering VoIP phones it causes a lot of problems because of a problems called "UDP fragmentation", when packets get too big to fit into a single UDP packet. Especially when a lot of codecs are presented by the phone, this limit can easily be reached. In that case it screws up the communication. There are many problems with UDP when the VoIP phone is behind a router or firewall, as there are several ways those devices behave in practice (symmetrical NAT) and how long those ports are kept open. Because of this unpredictable and erratic behavior, it is strongly recommended not to use UDP and use a connection oriented transport.
      TCP	The TCP makes sure that packets don't get lost between the client and the server. Many mainstream protocols such as HTTP and Email are using TCP for sending their packets back and forth, which make TCP an extremly robust and well-understood protocol in the Internet. The main drawback of TCP is that it is not encrypted. Some firewalls that are monitoring the traffic tend to change TCP packets, which can cause problems with the PBX. Today, most VoIP phones properly the TCP protocol for SIP, although it is not strictly mandatory according to the SIP protocol.
      TLS	TLS is based on TCP, but uses encryption. This makes it possible to negotiate secure keys for the voice packets as well, so that the whole call can be encrypted. The main problem with TLS is that the PBX should have a signed certificate. Those certificate can cost a significant amount of money and usually need to be renewed annually.

      By default, the PBX tries to provision TLS protocol, or at least the TCP protocol if the device supports it. Generally, this is the best chance for a seamless user experience.

      Unfortunately, it is not always possible to use TLS or even TCP. Some firewalls block TLS traffic because they are not able to analyse the traffic, and some other firewalls might not be able to handle too many persistent TCP connections so that UDP might become neccessary.

      PBX Address

      When a phone wants to send a registration to the PBX server it needs to have a resolvable address for the PBX. The SIP standard provides several ways to discover the PBX address. By default, the PBX uses the IP address method which works well in environments where there is no DNS available.

      Protocol	Description
      IP-Address	By default, the PBX determines what IP address the PBX needs to present to the VoIP phone. The way to figure out what IP address to present can be tricky; however the PBX has a reasonable way to determine this address which works well, even if the PBX is operating in a mixed IPv4/IPv6 address environment with multiple IP addresses.
      DNS-Address	Especially for hosted PBX setup where the provider is able to provide a DNS address for each domain, it makes sense to provision the domain name instead of an IP address. This makes it possible to use DNS SRV and NAPTR, especially for automatic failover from the client this is an easy way to provide multiple addresses where the phone can register. This method needs to be enabled by the setting "Use domain name instead of IP address" (Plug and Play Settings).

      Domain Outbound Proxy Pattern

      The Domain Settings may contain a special pattern that tells the PBX what address to provision for certain addresses.

      Along with the address, the PBX will also try to select the right port. This also depends on the transport layer, e.g. typically TLS has a different port than TCP or UDP.

      Using the Outbound proxy pattern, the outbound proxy that is provisioned into the phones can be controlled based on their public IP addresses. A typical scenario is that an office which is using a hosted PBX uses a SIP-aware firewall that needs to receive the outbound traffic from all IP phones in that office. The value in this field is a list of patterns that looks like this: src/mask/adr[:port[/transport]]. If the phone to be provisioned matches the source address src with the netmask mask, it will use the outbound proxy made of the adr:port and transport parameters. The port and the transport are optional. If not present, they will be replaced with 5060 and udp.

      So let's say that a tenant has two offices which have a VoIP router. The first office is at 1.2.3.4, and all traffic needs to be sent to 10.0.0.2. Then the first pattern would be 1.2.3.4/32/10.0.0.2. The second office would be in the VPN of the company and uses IP addresses 10.43.0.x, and the internal gateway is at edge43.company.com, using TLS on port 5061. Then the second pattern would be 10.43.0.0/24/edge43.company.com:5061/tls. In the domain setting the administrator would have to enter 1.2.3.4/32/10.0.0.2 10.43.0.0/24/edge43.company.com:5061/tls.

      Selecting the Transport Layer

      The transport layer can be pre-selected on extension level, on domain level and on system level. The setting is called "Proposed SIP transport protocol". Depending on the support of the VoIP phone and the provisioning template, the PBX will then try to provision the selected transport layer. If there is nothing set on the extension level, the PBX will try to use the domain level, and if that is also set to use the default value, it will use the system value, and if that is also on default it will attempt to use TLS, then TCP and as a last resort, UDP.

      If the Outbound proxy pattern was set on domain level it will override the transport layer selected on extension, domain or system layer.

      Changing the transport layer will take effect on the next provisioning event for the phones.

      Alcatel Phones

      Supported Devices and Features

      Vodia PBX currently supports provisioning Alcatel Temporis IP Phones : IP100, IP151, IP301G,IP251G, IP700, IP700G
      When using the automatic provisioning, some of the supported features are:

      • Basic call features including transfer, hold and voicemail notification
      • Status synchronization of the DND and redirection status
      • BLF mode for buttons
      • Time zone settings etc.

      Note: To reset a Alcatel Temporis phone to factory default settings, press the OK button during 20s

      Note: To reset a Alcatel Temporis phone to factory default settings, press the OK button during 20s

      Provisioning

      When provisioning a Alcatel phone, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning .

      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the registration tab as shown again below.

      

      It is also important to note in the document "Steps Before Provisioning", under "Configuring the Domain Settings", the "Provisioning Parameters" as shown again below.

      

      The Authentication user, Authentication password and Authentication PIN settings allow administrators to provision phones and log into a phone’s web interface remotely. These settings enable administrators to use identical usernames and passwords for all devices in a domain, simplifying the authentication process from the GUI of the phone. The password will bypass users’ passwords (the web interface passwords), so users will not be able to use their passwords; hence, this setting can be used to keep users out of the phone’s web interface and prevent them from inadvertently changing anything. The default username is admin, and the default password is password.

      For provisioning buttons, the setting up of buttons is described in buttons . There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension. The numbers for the buttons correspond to the buttons on the phone. If a number is not added in the profile, that button will not be set on the phone.

      Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC based provisioning" as shown, to allow the phone to provision to that specific account.

      

      Alcatel Temporis Settings for Auto Provisioning

      Log into the web interface of the phone and navigate to "Servicing", then "Provisioning" page.

      • In the "Server URL", enter the PBX URL http://PBXIP

      Then, reboot the phone.

      

      Cisco SPA Phones

      

      The PBX is able to automatically provision Cisco SPA devices through zero-touch LAN provisioning, DHCP option 66 and through manual entry of the provisioning URL in the phone.GUI.

      In order to use features like TLS and SRTP, you should upgrade the phones to firmware 7.6.2. This will also store the credentials for the provisioning in the phone, so that you don't have to open the MAC for provisioning every time that you want to reprovision the phone.

      Provisioning

      When provisioning a Cisco SPA phone, like SPA504G or SPA525G etc., or a Cisco ATA, like SPA112, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning

      Note: SPA504G or SPA525G require at least 2 line buttons to be provision on the phone.

      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the registration tab as shown again below.

      

      It is also important to note in the document "Steps Before Provisioning", under "Configuring the Domain Settings", the "Provisioning Parameters" as shown again below.

      

      The Authentication user, Authentication password and Authentication PINsettings allow administrators to provision phones and log into a phone's web interface remotely. These settings enable administrators to use identical usernames and passwords for all devices in a domain, simplifying the authentication process from the GUI of the phone. The password will bypass users' passwords (the web interface passwords), so users will not be able to use their passwords; hence, this setting can be used to keep users out of the phone's web interface and prevent them from inadvertently changing anything.

      For provisioning buttons, the setting up of buttons is described in buttons . There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension. The numbers for the buttons correspond to the buttons on the phone. If a number is not added in the profile, that button will not be set on the phone.

      Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC based provisioning" as shown, to allow the phone to provision to that specific account.
      

      Cisco SPA Phone Settings for Auto Provisioning

      You can manually tell the phone from where it should load the configuration either through the web interface of the phone or through the phone keyboard. If you want to use the phone keyboard, select the Profile Rule entry in the interface, though it seems to be much easier through the web interface.

      If you choose to use the web interface, log into the web interface of the phone, and according to the model, navigate to the page where you can enter the provisioning server address. In models SPA504G
      and SPA525G, the setting is called Profile Rule found on page "Admin" -> "Advanced" provisioning tab as shown below. In other models it could be on another page that needs to be found from the documentation of the respective model.

      In Profile Rulesetting, add this URL http://PBXIP/spa$MA.cfgwhere the PBXIP is the IP or resolvable domain of the pbx server.

      Make sure that Provision Enable (if present) is set to "yes".

      Press . Either the phone will reboot by itself or you can reboot the phone for provisioning.
      

      Setup Devices in LAN

      If your phone is in the same LAN as the PBX, then you don't have to do many of the above things and simply be able to provision almost any phone with a single press of a button by using this feature of the PBX. Make sure the phone is factory reset (or at least have the default username and password). You still have to do the Steps Before Provisioning but after that, just go to the PBX page Setup Devices in LAN where the PBX will show you all the phones it has automatically detected in the LAN. You simply choose the phones you want to provision, assign each one a domain and extension and press "Setup", and that's it. The phones will be provisioned. For details on this feature, please go to: Setup Devices in LAN.

      Cisco 3PCC Desktop Phones

      )

      The Vodia PBX is able to automatically provision Cisco 3PPC devices (68xx, 78xx and 88xx) through zero-touch LAN provisioning. This example is based on the 8841 model but will also work for Cisco phone 7800 and 6800 series.
      The devices should be running sip88xx.11-0-0MPP-7.loads or the latest build available .

      Upgrading Your Phone

      Make sure you host the firmware with an HTTP server or TFTP server, (One can find the tftp folder of the PBX inside /usr/local/pbx/tftp. If the folder is not found, please create a new one)

      and it looks like this URL http://ip_address_server_with_files/sip88xx.11-0-0MPP-7.loads. To prevent typos you could just copy paste the filename.

      Then submit the upgrade URL under Advance→ Voice →Provisioning setting field Upgrade Rule then save the changes

      

      Resetting Your Cisco 3PCC Series Phone

      • Remove power from the phone by unplugging the power adapter
      • Wait 5 seconds
      • Press and hold # and plug the phone back in.
      • When the Headset and Speaker buttons are lit, release # and enter the following key sequence: 123456789*0#

      Provisioning

      When provisioning a Cisco 3PPC desktop phone you should first set the required settings on the Vodia PBX as shown in provisioning document Steps Before Provisioning .

      The Authentication user, Authentication password and Authentication PINsettings, enable administrators to use identical usernames and passwords for all devices in a particular domain, simplifying the authentication process from the GUI of the phone when the phone is provisioned to the Vodia PBX.

      

      Setup Devices In LAN

      If your phone is in the same LAN as the PBX, you can provision the Cisco 3PCC devices by visiting the (Phones in the LAN ) under the Administrator level (Phones→LAN Devices ).
      Make sure the phone is factory reset (or at least has the default username and password).
      

      Setup The Extension

      Make sure Steps Before Provisioning is complete before proceeding to provision. Continue and choose your Cisco 8800 phone you want to provision, assign each one a domain and extension and press "Setup", and that's it.

      The Cisco 8800 will reboot and then be provisioned to the Vodia PBX.

      

      Provisioning Buttons

      For a full description visit (buttons). There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension.
      The numbers for the buttons correspond to the buttons on the phone.

      

      Remote provisioning

      If you choose to use the web interface, log into the web interface of the phone, and according to the model, navigate to the page where you can enter the provisioning server address. In models SPA504Gand SPA525G , the setting is called Profile Rule found on page "Admin" -> "Advanced" provisioning . In other models it could be on another page that needs to be found from the documentation of the respective model.
      In Profile Rulethe setting, add this URL http://PBXIP/csc$MA.cfg where the PBX IP is the IP or resolvable domain of the PBX server. The phone will auto provisioning without having to reboot it.

      Grandstream Phones

      ---

      

      Provisioning Grandstream Desktop Phones

      When provisioning a Grandstream phone, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning .

      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the registration tab as shown again below.

      

      It is also important to note in the document "Steps Before Provisioning", under Domain>Advanced> General Settings> Provisioning Parametersas shown again below.

      

      The Authentication user, Authentication password and Authentication PIN settings allow administrators to provision phones and log into a phone’s web interface remotely. These settings enable administrators to use identical usernames and passwords for all devices in a domain, simplifying the authentication process from the GUI of the phone. The password will bypass users’ passwords (the web interface passwords), so users will not be able to use their passwords; hence, this setting can be used to keep users out of the phone’s web interface and prevent them from inadvertently changing anything.

      For provisioning buttons, the setting up of buttons is described in buttons . There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension. The numbers for the buttons correspond to the buttons on the phone. If a number is not added in the profile, that button will not be set on the phone.

      Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC based provisioning" as shown, to allow the phone to provision to that specific account.

      

      Grandstream Phone Settings for Auto Provisioning

      Log into the web interface of the phone, and according to the model, navigate to the page where you can enter the provisioning server address. In model GXP2200, the setting is called Config Server Path found on page "Maintenance" -> "Upgrade" as shown below. In other models it could be on "Upgrade" and "Advanced" page or something else that needs to be found from the documentation of the respective model.

      In "Config Server Path" setting, add this URL PBXIP/prov where the PBXIP is the IP or resolvable domain of the pbx server.

      Make sure that "Upgrade Via" (if present) is set to "HTTP".

      PressSave. Now you can reboot the phone for provisioning.

      

      Setup Devices in LAN

      If your phone is in the same LAN as the PBX, then you don't have to do many of the above things and simply be able to provision almost any phone with a single press of a button by using this feature of the PBX. Make sure the phone is factory reset (or at least have the default username and password). You still have to do the Steps Before Provisioning but after that, just go to the PBX page Setup Devices in LAN where the PBX will show you all the phones it has automatically detected in the LAN. You simply choose the phones you want to provision, assign each one a domain and extension and press "Setup", and that's it. The phones will be provisioned. For details on this feature, please go to: Setup Devices in LAN .

      GRP2612/13/14

      Vodia PBX 63.1 and higher supports Grandstream GRP2612/13/14 models. The provisioning is the same like with the other models, with the following differences:
      LAN provisioning: Because the Grandstream phones now use a device-dependent password for the web interface, the zero-touch does not work any more. In order to have this working, you now need to log into the web interface of the phone and use the password printed on the back of the phone to set the config server path to the path to the PBX. Cloud provisioning using Grandstream RPS should still work the same. For the LAN provisioning follow the steps:

      1. Add the phone MAC address to the extension and make sure that the extension is opened up for provisioning/pairing with the phone.
      2. Power the device up and look up the IP address of the phone. This can be done through the keyboard of the phone.
      3. Log into the web interface of the phone. The username is "admin" and the password is printed on the back of the device. The PBX will provision the domain admin password into the phone after the phone was configured the first time.
      4. Navigate to "maintenance" and "upgrade and provisioning". Set the "Config Server Path" to the address of the PBX, e.g. 192.168.1.2 and make sure that the "Config Upgrade via" is set to HTTPS.
      5. Press the "save and apply" button. The phone should go ahead and fetch the configuration from the PBX. You will see a symbol on the screen of the device indicating that it is pulling the configuration from the network.
      6. After the phone received its original configuration, you can go ahead and change the button configuration or make other changes to the configuration.

      Buttons: The phones need at least one private line for the buttons. If there is no private line, the phone may reject the entire configuration.

      Htek Phones

      

      Supported Devices and Features

      Vodia PBX currently supports provisioning Htek phones UC926, UC924, UC862, UC842, UC803P, UC802P.
      When using the automatic provisioning, some of the supported features are:

      • Basic call features including transfer, hold and voicemail notification
      • Status synchronization of the DND and redirection status
      • BLF mode for buttons
      • Time zone settings etc.

      Note: To reset a htek phone to factory default settings, go to the main menu, then go to Settings, then Advanced Settings, enter the web password, then Phone Settings, then press Factory Reset and confirm by pressing ok.

      Provisioning

      When provisioning a Htek phone, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning .

      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the registration tab as shown again below.

      

      It is also important to note in the document "Steps Before Provisioning", under "Configuring the Domain Settings", the "Provisioning Parameters" as shown again below.

      

      The Authentication user, Authentication password and Authentication PIN settings allow administrators to provision phones and log into a phone's web interface remotely. These settings enable administrators to use identical usernames and passwords for all devices in a domain, simplifying the authentication process from the GUI of the phone. The password will bypass users' passwords (the web interface passwords), so users will not be able to use their passwords; hence, this setting can be used to keep users out of the phone's web interface and prevent them from inadvertently changing anything. The default username is admin, and the default password is password.

      For provisioning buttons, the setting up of buttons is described in buttons . There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension. The numbers for the buttons correspond to the buttons on the phone. If a number is not added in the profile, that button will not be set on the phone.

      Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC based provisioning" as shown, to allow the phone to provision to that specific account.
      

      Htek Phone Settings for Auto Provisioning

      Log into the web interface of the phone and navigate to "Management", then "Auto Provision" page. In the "Config Server Path", add this URL http://PBXIP
      You don't have to reboot the phone for provisioning, it is okay to save the information and then use theAutoprovision Now button.

      

      Poly phones

      

      Supported Devices and Features

      The Vodia PBX currently (version 60.0) supports provisioning Polycom phones VVX101, VVX201, VVX300, VVX301, VVX310, VVX311, VVX400, VVX401, VVX410, VVX411, SPIP300, SPIP301, SPIP320, SPIP330, SPIP321, SPIP331, SPIP335, SPIP430, SPIP450, SPIP500, SPIP501, SPIP550, SPIP560, SPIP600, SPIP601, SPIP650, SPIP670, SSIP4000, SSIP5000, SSIP6000 and SSIP7000.

      When using the automatic provisioning, you will be able to make and receive calls, transfer calls, control the DND and redirection status on the PBX, use the Polycom buttons (restrictions apply) and access the PBX address book through LDAP. LDAP uses TLS so that the address book credentials and the address book information stays secure between the phone and the PBX. The PBX can use TLS transport layer for the SIP messages, eliminating NAT problems and problems with UDP message fragmentation due to the Problems with UDP on Polycom phones. If TLS cannot be used, it is strongly recommended to use at least TCP.

      If the domain contains a logo, the PBX will provide that logo to the phone as the background image. The logo takes up by default a size of 50 % of the screen, which provides enough space for the button information on the phone. For the background color, dark color seem to work better for the phones. Consequently, if a domain is using mostly Polycom phones, it is suggested to choose a dark background color when uploading the logo into the domain.

      Polycom VVX 300/310/400/410 Reboot & Factory Reset

      Pressing multiple key combinations enable you to perform certain operations on the VVX 300, 310, 400, and 410 phones. The multiple key combinations on theVVX 300, 310, 400, and 410 phones are as follows:

      • Reboot the Phone—Press and hold the dial pad keys 0, 1, and 3 simultaneously for about three seconds , or until you hear a confirmation tone.
      • Restore Factory Defaults—Press and hold the dial pad keys 1, 3, and 5 simultaneously during the Updater process until the password prompt appears. Enter the administrator password to initiate the reset. Resetting to factory defaults will also reset the administrator password to 456.
      • Upload Log Files—Press and hold the dial pad keys 1, 5, and 9 simultaneously until you hear a three­second confirmation tone.

      Lost the Administrator Password?

      To restore a third-party phone to factory defaults (or when you otherwise can't access the Admin Password):

      • Power cycle the phone:Simultaneously press and hold 0 1 3 dial pad keys * until a confirmation tone is heard (or for about three seconds).

      • Except VVX 1500: 2 4 6

      • Press Cancel soft key during the boot process.

      • During the countdown, simultaneously press and hold the 1 3 5 dial pad keys until the password prompt appears.

      • Enter the MAC ID of the phone. (The MAC ID/address should be visible on a label on the underside of the phone.)

      Note: When entering the MAC ID of the phone use lower case for the letters.

      Preparing Secure Provisioning

      The PBX will attempt to use secure provisioning for Polycom devices through HTTPS if the setting Try to provision phones securely is set. By default, the PBX will provision the Vodia Root CA into the phone, so that a trust relationship can be established. Instead the administrator can upload a certificate chain into the PBX; in this case, it is important that the whole chain including the Root CA is imported so that the PBX is able to provision the Root CA into the phone. This step is not needed when the PBX is using a certificate that is on the list of trusted Root CA for Polycom phones.

      Establishing a secure connection is also required for using TLS for the SIP signaling messages. If you cannot establish a secure connection, you should assign the TCP transport layer to the device.

      Providing the Firmware

      The firmware for the Polycom phones can be updated through the web interface of the phone or must be downloaded from the Polycom web site. You may use the tftpfolder for storing the firmware files.

      Supported Button Modes

      Polycom phones can use the BLF-based modes that monitor accounts (such as Extension BLF, ACD, Hunt groups or service flags). Speed dial is also supported; this has to be done through the Polycom address book which is automatically generated for the PBX. Park orbits can also be monitored on the Polycom phones.
      Because of the way the Polycom phones work, it is at times difficult to exactly position the buttons on the device. When there is a display next to the button, users should use that display to confirm the button functionality. It is important that you assign at least one line button to the button profile; otherwise, the phone will not accept the BLF configuration.

      Supported Transport Layers

      Polycom phones may use TLS, TCP and UDP transport layers. It is recommended to use the TLS transport layer unless there is a problem provisioning the certificate information for the phone. UDP is discouraged, as it causes problems with large packets and UDP fragmentation.

      Preparing the Device

      Before you can provision a phone, you must add the MAC address to the extension in the PBX web interface. This can be done either by adding the MAC address to the extension or by assigning the MAC address to an extension on the system level (for LAN provisioning).
      You need to tell the phone the provisioning server (where it will get its settings and firmware). For this, you have several possibilities:

      LAN discovery

      Manual Setup

      After the phone booted up, you can change the settings for the provisioning server from the phone user interface:

      • Press the menu button.
      • Select Settings (3)
      • Select Advanced (2)
      • Enter the password (by default 456)
      • Select Admin Settings (1)
      • Select Network Configuration (1)
      • Select the Server Menu
      • Set the Server Type to "HTTP"
      • Set the Server Address to the address of the PBX. If you are deploying the device in a hosted environment, you need to enter the name of the domain here; otherwise, the IP address is sufficient.

      Manual Provisioning:

      When provisioning a Polycom phone, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning .
      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the registration tab as shown again below.

      

      It is also important to note in the document "Steps Before Provisioning", under "Configuring the Domain Settings", the "Provisioning Parameters" as shown again below.
      

      The Authentication user, Authentication password and Authentication PIN settings allow administrators to provision phones and log into a phone's web interface remotely.

      For provisioning buttons, the setting up of buttons is described in buttons. There, you will see how to prepare a new buttons profile, how to set up the individual buttons (like BLF) and how to map that profile to a specific extension.

      P.S: There should be at least one " Private Line" button assigned in the button list of the Polycom phone which is about to be provisioned, for the phone to register. Otherwise, the phone will not register.

      Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC-based provisioning" as shown, to allow the phone to the provision to that specific account.
      

      Polycom Phone Settings for Auto Provisioning

      Log into the web interface of the phone and navigate to "Management", then "Auto Provision" page. In the "Config Server Path", add this URL http://PBXIP
      after the URL has been set in the provisioning page, you can restart the phone to start the process.

      

      Using the DHCP Server

      When you have a DHCP server then can set option 66, you can just enter the address of the PBX there. If you are deploying the device in a hosted environment, you need to enter the name of the domain here; otherwise, the IP address is sufficient. Please make sure that the setting is an HTTP URL.

      Yealink Phones

      

      Supported Devices and Features

      Vodia PBX supports provisioning Yealink phones CP960 T31P CP920 T19P T20P T21P T22P T23G T23P T26P T27G T27P T28P T29G T30 T30P T31 T31G T32G T33G T33P T38G T40G T40P T41P T41S T41U T42G T42S T42U T43U T46G T46S T46U T48G T48S T48U T49G T52S T53 T53W T54S T54W T56 T57W T58 VP59 W41P W52P W53P W56P W60B W60P W80B

      When using the automatic provisioning, some of the supported features are:

      • Basic call features including transfer, hold and voicemail notification
      • Status synchronization of the DND and redirection status
      • BLF mode for buttons
      • Time zone settings etc.

      Factory Default a Yealink Phone

      To reset a Yealink phone to factory default settings, hold down the "OK" button for 5-10 seconds. The phone will prompt you to reset to factory default, press OK. For devices that don't have an "Ok" button like the T58 model, use the redial button.

      Manual Provisioning

      When provisioning a Yealink phone, you should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning.

      It is important to note in the document "Steps Before Provisioning", how to map the MAC address of the phone to the PBX extension under the provisioning tab as shown again below.

      Step 1: Add the MAC address in the extension

      

      Step 2: Add password and the PIN

      • It is also important to note in the document "Steps Before Provisioning", under "Domain Settings", the "Provisioning Parameters" as shown again below.

        

      • This is the password which will act as web password to get into the GUI of the phone, AFTER it provision

      • The default username is admin , and the default password is password .

      • For provisioning buttons, the setting up of buttons is described in buttons.

      Step 2: Open MAC Provisioning

      For security reasons, navigate to the extension level and activate "Open account for MAC based provisioning" as shown, to allow the phone to provision to that specific account.
      

      Step 3: Steps to be carried out on Yealink's GUI for manual provisioning.

      • Log into the web interface of the phone and navigate to "Upgrade" > "Advanced". In newer models, it may "Settings" and "Auto Provision". In provisioning server, add the URL http://PBX_IP_OR_Domain_name.

      • Make sure that "Check New Config" is set to "Power on".

      • You don’t have to reboot the Yealink phones for provisioning, it is okay to save the information and then use the Autoprovision Nowbutton.

      

      ---

      Yealink RPS / Auto Provisioning

      • Step 1: Make sure the PBX is publicly accessible.

      • Step 2: Get the API username and the API Key from the Yealink's RPS server's section to enter it on the PBX as shown below.

        

      • Step 3: Make sure that the phone is factory reset before going further.

      • Step 4: Enter the mac address of the phone in the extension created on the PBX and hit Save

      • Step 5: The phone needs to reboot. If it doesn't do it by itself, manually reboot.

      • Step 6: The MAC address along with the domain name of the PBX should now be seen populated in the RPS server. The phone should be fully provisioned and registered now.

        Gigaset Phones

        Maxwell devices

        When provisioning a Gigaset phone, we should first set the required settings on the PBX as shown in provisioning document Steps Before Provisioning. The provisioning then is like provisioning every other supported phone.

        The PBX is able to automatically provision Gigaset phone devices through zero-touch LAN provisioning, DHCP option 66, cloud provisioning throgh the Gigaset redirection server and through manual entry of the provisioning URL on the phone GUI. The redirection service will require that you enter the PIN for the MAC address, which is on the back of the device near the MAC address.

        Resetting Maxwell devices

        
        How to Factory Reset Gigaset Maxwell

      • Find the phone IP address.
      • Press the Control key when the phone is idle, navigate to Settings → Info on the phone.
      • Log in the phone web interface with your password: admin.
      • Go to System → Reboot and Reset, click Reset to Factory Settings .

      N670, N870 DECT

      In order to set up the N670 and N870, you'll need to assign the extensions with the MAC address of the base station that will pull the configuration. This can be done by adding the MAC to each extension and then setting the parameter 1 for the extension. If there are many extensions, the following example may serve as template for importing the data in one step and assigning it to the extensions:

      #alias;mac;param1
      40;7C2F80001122;029e74a598
      41;7C2F80001122;029e74a599
      

      Text

      In order to start provisioning, at least one of the extensons needs to be ready for provisioning device (open accounts for MAC-based provisioning). The PBX expects an URL with the path 83/1/provisioning.xml in the provisioning URL, which is the default for the N670 and N870.

      Avaya Phones

      Supported devices and features

      Currently (version 68.0.22) the devices of the J100-series are supported (J129, J139, J159, J169, J179, J189).
      Most of the features are supported:

      • Automatic provisioning
      • Automatic sync of the provisioning files
      • Firmware provisioning through parameters
      • Secure provisioning and secure calls (SRTP)
      • Generic parameter for controlling other settings
      • Calling, transfer, parking of calls
      • Buttons assigments including speed dial and BLF
      • LDAP integration
      • Multicast paging in the LAN
      • Background images
      • Custom ring tones

      Important limitations are the missing synchronization of DND and redirection settings with the server (as-feature-event). Also, there is no mini-browser available at this time.

      Installation

      It is strongly recommended to download the supporting files from the Avaya web page (including the firmware and language files) and drop them into the tftp folder of the PBX. The phone should then pick up the firmware automatically.
      This description is based on version R4.0.10.2.2, but Avaya may release newer versions with further improvements.
      If you are using the Avaya redirection service, please make sure that you open the phones up for provisioning, so that they can fetch the credentials for futher provisioning.
      If you can not use the Avaya redirection service, you can use secure provisioning if your PBX has a valid certificate. Again, it will be important to open up the extension for provioning after adding the MAC address to the extension and before starting the phone. For manual provisioning, you hae to make sure that the web server is enabled. Even if there is no valid certificate on the phone, use a secure HTTP connection to the phone (e.g. by telling the browser that the connection is secure). Otherwise the browser might not be able to send the login credentials to the phone. The default username is admin and password is 27238.
      After provisioning the device, the web server will be enabled. The username and password for the phone is the tenant provisioning username and password like for all other supported phone manufacturers.

      Background images

      The PBX will automatically generate reasonable images for the background of the screens. It uses the dark mode. This means that companies logo that use the white color will look best on the screen. It looks good to use a solid background color, but background patterns are also looking good on the phones.
      The logo itself will be placed at the top middle of the screen where it does not interfere with other dialogs, e.g. when searching a contact. The idle screen uses a transparent version of the logo so that it does not interfere with the time too much. The image for the secondary screen does not include the logo. Only the idle image for the secondary screen contains a full-size logo. If the customer wishes a different layout, custom images can replace the four automatically generated images. Please make sure that they are in the 320 × 240 pixel size JPEG format.

      Custom ring tones

      The PBX will automatically provision custom ring tones like with all other supported devices. You can upload your own ring tones through the PBX web interface. If you want to use the Avaya built-in ring tones instead, you can set the parameter avaya-custom-ringtones to 0. In this case, the PBX will still use the Alert-Info headers for controlling which ring melody will be played for incoming calls.

      Button assignments

      The PBX uses the line mode, the autodial mode and the BLF mode of the Avaya phones. That means that private lines, speed dial and account-based functions like BLF, park orbits or queues can be used.
      Additional modes can be provisioned with the Avaya mode, where the parameter is passed through to the provisioning file. This makes it possible to use every possible mode for the Avaya phones, albeight the configuration is lacking convenience.

      IPv6 and network settings

      The PBX is able to support IPv6, however does not interfere with the settings on the phone regarding IPv6. If the network is configured properly, IPv6 should work seamless.
      The PBX configuration template for the Avaya phone does also not st anything up regarding QoS settings or VLAN. If there is a need to configure anything from the template, the general parameter can be used for this purpose.

• snom360: http://wiki.snom.com/Firmware/Update/TFTP_Update

Types of Manual Provisioning of the Snom phone

Each method of plug and play is described below. Before doing Plug and Play though, take a look at the pre-requisite steps .

• DHCP, option 66 : This method can be used to provision in both LAN (internal) and WAN (remote) environments as long as you can control the local DHCP server and that DHCP server supports option 66. The phone retrieves its IP address and other configuration information through dynamic host configuration protocol (DHCP). In order to use this method, you need to put in the address of the PBX, either the IP-address or the DNS-address, along with the HTTP transport scheme (for example, "http://192.168.1.2 " or "http://pbx.abc.com ").

• LAN provisioning : If your PBX is in the LAN and your phones are in the same LAN, then the PBX is smart enough to detect the make and the model of the phone (along with other efficient details) to present the user with an option of "setting up" the Phone to provision with a few clicks.

1. If the "Setup" button doesn't come up (as above) that may mean one of the following:

• PBX or the Phone is not on the same LAN as the other.
• The MAC address that it detected is already being used by an extension under some domain.
• Extension needs a factory reset to be fully functional at this point.

2. If the above condition is satisfied, then the setup button appears and you can now assign that MAC address to an extension under a domain of your choice.
3. This process handles the opening of the MAC Based provisioning, assigning the MAC address to the desired extension and actually provisioning the phone all by itself.

• Manual setup : This method is done by accessing the phone through a web browser. You will need to configure the phone’s Settings URL setting so that the phone knows where the Vodia PBX is.

1. The settings URL for the "Update " section of the Snom phone is: " http://pbx_domain_or_ip_address "
2. Open the "MAC Based Provisioning" setting by selecting the extension.
3. You can even assign the buttons to the individual phone or for the whole domain from under Domain> Advanced> Buttons
4. Save this setting and "Reboot" the phone.

Auto provisioning the Snom phones (Needs a Hosted License only! )

With this PBX feature, you can easily redirect your snom phones to your Vodia PBX PBX for provisioning via the SRAPS server. It makes provisioning of Snom phones very easy. Ideally, all the user has to do is plug in his/her phone and he should be good to go. The phone should get all the settings, be registered and its buttons set, according to how they had been set up in the PBX.

Steps:
1.

Enter Username and Password for your SRAPS server here and save it.

2. Redirecting a Phone to the SRAPS server (Auto Provisioning)

After you have done the above step, you can redirect your snom phones. To redirect a snom phone to your Vodia PBX for provisioning, just add the MAC address of the phone (found on the phone label at the back) to the PBX:

• Go to your selected domain in Vodia PBX web interface.
• Go to the extension to which you want to bind your snom phone.
• Go to the Registration tab of the extension found at the top.
• Enter the MAC address of your snom phone in Bind to MAC Address setting.
• Press Save .

This will not only bind the extension to that phone’s MAC address, it will also redirect that phone to your PBX when it requests for provisioning and will be provisioned by the PBX accordingly.

Syncing All MACs on the System

Generally, you won’t need this because when you add a MAC address for binding, it also automatically registers it with the snom provisioning server for redirection, according to the credentials and PBX address you have provided in the PnP Settings.

But if there were any MAC’s present on the system before you entered the PnP Settings for the Provisioning Server, those MAC’s would not have been redirected. For that or any other reason that some mac addresses are not redirected, you can redirect all the MAC’s on the system with a single press of the Sync button. It is found on the same page where you entered the provisioning server credentials.

Defining Which Firmware Should be Used

There have been many enhancements starting version 58.0. One of them was to subsidize the PNP section into text only. Today if you would like to change the firmware version, you will have to switch to admin-mode then navigate to Parameters, on the Type box edit the Firmware you would like to upgrade or downgrade to. You can see the URLs that correspond to each phone, you can change the link there or change the version number for example 8.4.35 to 8.5.36

Manual Provisioning

Steps :

1. Add MAC address by clicking on the "+" button under "MAC " section under Extension's list.

2. Enter the Manual provisioning URL as: "http://PBX_IP_OR_Domain_name"

3. Select the Extension to open it up for MAC based Provisioning.

4. You can even assign the buttons to the individual phone or, to the whole domain from under ```
   Domain> Advanced> Buttons.

5. Save this setting and "Reboot" the phone.

6. Reboot the phone.