ScopServ - Basic Installation Hierarchy for Telephony Server New Design v04_2.pdf

ScopTELTM IP PBX Software


Basic Installation Hierarchy for Telephony Server


Basic Installation Hierarchy for Telephony Server

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Objects like telephony extensions, Class of Service, and Outgoing Lines require that other objects existbefore they can be properly and efficiently configured.

Configuring these objects in the correct sequence will make first time installation much more efficient andpromote a better understanding of each object’s intended function.

Therefore the purpose of this document is to provide a visual walkthrough of a very basic but functionalinstallation for one tenant. This tutorial does not include an overview of the overall network configurationbut does cover a basic Telephony Firewall Configuration.

IMPORTANTE NOTE: It is best practice not to use Upper Case characters or Special characters when entering object names.

Here is a basic checklist to perform when setting up a new server for the first time.

1. Edit Telephony Modules2. Edit Channels3. Edit Feature Codes4. Commit5. Edit Services Startup6. Packages Manager7. Interfaces Card Detect (optional but at least one physical interface must exist)8. Create VoIP Account (optional but at least one physical interface must exist)9. Create Group Interface(s)10. Create Outgoing Line(s)11. Create Class of Service objects12. Security Settings13. Create Extensions14. Create Incoming lines15. Automatic Provisioning Service16. Commit and Services Restart


Basic Installation Hierarchy for Telephony Server Provisioning Pre‐Requisites

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• A TFTP Server must be running on network to provision supported IP Phones.

• Make sure that TFTP Service is running on the ScopTEL server under Server General configuration tab.

• If the ScopTEL DHCP server is being used then ensure that under Advanced Options tab that TFTP Server name is configured to point DHCP clients to LAN IP address of ScopTEL server. Also ensure only one DHCP server is servicing that subnet. Restart the DHCP Service if any changes are made to the DHCP Server’s configuration.

• If a third party DHCP server is being used on network then ensure that DHCP server on network uses Option 66 (TFTP) and points DHCP clients to LAN IP address of ScopTEL server. Also ensure only one DHCP server is servicing that subnet.

• Reboot any IP Phones connected to the network so they can download their configurations.• If you are unsure that the phones are downloading their configurations there are two troubleshooting methods to verify

configuration downloads are working.

1. Navigate to Tools > File Manager > TFTP Directory and inspect each configuration file for proper configuration values. Also put any required firmware update files into this directory if any phones require to have their firmware updated.

2. From a root CLI session you can monitor network traffic on TFTP port 69 with the following command (omitting all quotation marks) “ngrep –d any port 69”. If DHCP is properly pointing client TFTP requests to the ScopTEL server then from this CLI session you can see any file requests being received from the IP Phone during it’s boot process. Verify the filename’s are correct if you see TFTP traffic reaching the ScopTEL server.


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1. Edit Telephony Modules

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It is essential that pre‐requisite modules be enabledprior to anything being configured on theserver. This is to ensure that only required VoIPprotocols and PSTN hardware modules areloaded during startup.

Edit and save the values for tab Configuration >Telephony > Configuration > TelephonyModules. Check off only the values you require.

Recommended values are default values and AnalogInterfaces (Zaptel), Digital Interfaces (Zaptel),SIP, IAX, Virtual Fax (requires Hylafax,IAXmodem, packages and relateddependencies), Emergency Lines, Queues andAgents, Conferences, IVR Report/Logging,Scheduler, Asterisk Manager.


You must edit your voice channels before you can finish committing your configuration. Configuration >Telephony > Configuration > Channels

This is a good time to configure your preferred CODEC order based on your geographical region. Forexample in North America the default CODEC used by carriers for T1 and SIP voice channels isG.711 MuLaw. End points normally negotiate CODEC selection with other end points based onthe first compatible CODEC listed in the preferred CODEC order.

Note that server support for the G.729 CODEC is not possible without appropriate third party G.729licenses or CODEC transcoding hardware. Transcoding is very CPU heavy therefore hardwaretranscoding is recommended if necessary since this offloads transcoding from the ScopTEL CPU.

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2. Edit Channels

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Each feature of the Telephony Server is activatedby the user using DTMF feature codes.When the server recognizes DTMF featurecodes entered via a user’s extension then theserver verifies this feature request using the“Class of Service” configured for thatextension. If the feature is not listed in theClass of Service configured for that extensionthe server denies the request and the “CallFails”.

The feature code list is basically a list of DN’s(Directory Numbers). This list may becustomized but the feature code must notmatch another Application, Extension,Outgoing Line, Emergency Line, or SpecialLine. In order to emulate “En Bloc” signallingeach phone provisioned by the ScopTEL’s APS(Automated Provisioning System) mustinclude fixed dial plan options/entries foreach feature code.

Edit Configuration > Telephony > Configuration >Features Code

3. Edit Feature Codes


Once the modules, channels, and feature codes have been edited it isnecessary to commit the changes to the server so that theproper service start‐up options can be configured and started.

Any time a change in configuration is detected the “Commit” buttonwill be visible at the top right hand corner of the ScopTELTelephony GUI module screen. Click this button to reload allconfigurations after you have saved your required changes.Clicking on this Commit button will only write detected changesto the internal database configurations.

If a full configuration re‐write is required then a “Full Commit” can bedone from Configuration > telephony > Configuration > Generalscreen. The “Full Commit” is only possible from this screen.Note: This Commit button appears as a normal “Commit” buttonat the top right corner of the Telephony GUI page. Thisscreenshot displays a “Full Commit” from the “General” tab.

4. Commit


Configuration > Telephony > Configuration > General > Edit Services > Edit

Check off the services required after a server rebootso they start automatically without manualintervention and so that they are loaded inthe correct order. Apply Changes when youare done to return to the “General” screen.

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5. Edit Services Startup


If a service cannot “Commit” or cannot startbecause it is not installed then you mustinstall the required packages anddependencies.

A valid software maintenance contract mustbe in place with ScopServ in order toupdate or install ScopTEL packages. Toinstall any required packages navigateto Configuration > Server > PackagesManager > Version Information andclick on any required links to install anyrequired packages

6. Packages Manager


If any analog FXO/FXS or T1/E1 or BRI cards are installed then you must do a “Card Detect” to recognize and configure that hardwarebefore the drivers and configurations can be properly loaded. Configuration > Telephony > Interfaces > Detect Cards

Follow the pop‐up windows to complete the card detection procedure and be certain to read and follow any instructions that willappear in those pop‐up windows. After your PSTN hardware is detected and the required services are running it will be necessaryto configure regional properties and gain settings for each of your PSTN cards and ports. If a change is made to any settings on the“Interfaces” tabs it is a good practice to “Commit” those changes and then restart the following services in the correct order. Firstnavigate to the “General” tab…

The correct order to reset services is:

‐Stop the “Telephony Server” ‐Restart the “Analog/Digital Modules (Zaptel/Wanpipe) Service ‐Start the “Telephony Server”

7. Interfaces Card Detect


A typical VoIP Interface would normally use the industry standard SIPtechnology. Therefore this example will cover the creation of aCPE side (Customer Provided Equipment) SIP account.

SIP trunks normally require a SIP Registrar for client authentication anda SIP Proxy to handle signaling and media. It is common for theSIP Registrar and SIP Proxy to be the same server but the SIPRegistrar and SIP Proxy can reside on different servers.

To create a new SIP Interface navigate to Configuration > Telephony >Configuration > Interfaces > VoIP Accounts and Click “Add a NewVoIP Account” and choose SIP from the drop down list.

8.0 VoIP Accounts


The name field is mandatory and it is typical for thename of the SIP trunk to equal the SIP usernameprovided by the SIP ITSP (Internet TelephonyService Provider). A SIP Friend allows bothIncoming and Outgoing Calls and is most typical.

In this example the name of our SIP trunk is5555551212. When the General properties areentered click on the Server tab next to add theauthentication properties and the address of theSIP Registrar and Proxy.

8.1 VoIP Accounts


Authentication Mode is normally Plaintext but the username and password aresent over the Internet using MD5 encryption. Authentication Mode = MD5requires the MD5 hash entered into the text box and is not normally used.Enter the Username and Password into the corresponding fields in the GUI.

Host Mode Specific is chosen when the server is authenticating to a SIP serviceprovider. This is the option chosen to authenticate to an ITSP. This optionrequires that the ITSP’s hostname or IP address is entered in the text field.Host Mode Dynamic is chosen when another SIP user agent is registeringinbound to the server. Therefore the hostname and IP address can be leftblank.

In this example we are authenticating outbound therefore we must choose HostMode specific. If Registration must be forced then check the box forRegister as User Agent. When finished here click on the Network tab

8.2 VoIP Accounts


If the server is situated behind a third party NAT Router/Firewall then check off thebox for Trunk Behind NAT.

If the server is situated behind a third party NAT Router/Firewall then the thirdparty Firewall rules must port forward the following rules to the LAN IPaddress of the ScopTEL server. 5060/udp 10000‐20000/udp.

Also a static public IP address and/or FQDN (Fully Qualified Domain Name) isrecommended for the ScopTEL server to help negotiate Firewall related RTPissues. This IP address or FQDN is entered into the text field located atConfiguration > Telephony > General > External IP or Hostname and the“Server behind NAT ?[ ]” option be checked [x].

It is common for most SIP ITSP’s to require the Insecure options Port, Invite to bechecked.The SIP Qualify Time is defined in seconds (default 300 seconds).The server will send a SIP qualify message to the ITSP every 300 seconds. Thisallows the ITSP to monitor the status of the SIP registration and latency.When finished here click on the Options tab.

8.3 VoIP Accounts


The correct DTMF mode must be selected to matchwith the requirements of the ITSP. Typically“Automatic RFC‐2833/inband” is selected.However most ITSP’s support RFC‐2833because it is more reliable.

The CODEC selection must match the requirementsof the ITSP and any supported CODEC’s notchecked off will never be negotiated with theITSP because they have not been allowed. Theglobal Channel options defined inConfiguration > Telephony > Configuration >Channels > Codecs will choose the firstsupported CODEC in the preferred order.

No other options are required at this time so it is OKto click on the “Add” button.

8.4 VoIP Accounts


Once your PSTN hardware is detected and the required services are running you can set up Interface Groups.

An Interface Group is a “pool” of physical interfaces.

Configuration>Telephony>Configuration>Interfaces>Interface Group

The Group Interface can be a collection of PRI, FXO, SIP, or other technology interfaces but it is normally a collection of only onetechnology.

The purpose of an Outgoing Line Group is to isolate outgoing physical interfaces to specific Applications, Extensions, Outgoing Lines,Emergency Lines, Special Lines

For example there are 10 FXO (analog PSTN lines aka POTS lines) ports shared between two companies.FXO ports 1‐2 belong to Company ABCFXO ports 3‐4 belong to Company XYZGroup 1 is a collection of FXO ports 1‐2Group 2 is a collection of FXO ports 3‐4Therefore Group 1 belongs to Company ABC and Group 2 belongs to Company XYZ.

Later when the Outgoing Lines are configured those Outgoing Line prefixes can be configured to use either Group 1 or Group 2.

Now these new Outgoing Line objects can be assigned to unique Class of Service objects which are then later assigned to extensionsbelonging to either Company ABC or Company XYZ.

Therefore when an extension with an assigned Class of Service allowing access to Group 1 dials the outgoing line prefix 9 the extensionwill be allowed to dial from ports 1‐2

Therefore when an extension with an assigned Class of Service allowing access to Group 2 dials the outgoing line prefix 9 the extensionwill be allowed to dial from ports 3‐4

Both companies use the same outgoing line prefix 9 but cannot inadvertently dial out and incur expenses to the other company.

The outgoing ANI is also displayed correctly for each company.

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9.0 Interface Groups


To set up a new Interface Group navigate to Configuration > Telephony > Interfaces >Interface Group then Click Add a new Group

Choose a numeric Group number. Description is optional but recommended. Dial Mode isnormally in descending mode when the server is in a CPE environment to prevent“Glare”. Glare is when an incoming interface is ringing and an outgoing channel ringsimultaneously and collide with one another. This leads to an outgoing user answeringan incoming call unintentionally. Therefore if the incoming channels are ascending theoutgoing channel selection should be in descending order. Click on the “Add” button tofinish creating the Interface Group.

9.1 Interface Groups


Outgoing Lines use an outgoing prefix to select a physical Interface or Interface Group. The Outgoing Lineprefix cannot conflict with the leading digit of any Application, Feature Code, or Extension. It is fine ifthe leading digit of an incoming DNIS matches the leading digit of an Outgoing Line because theIncoming Line context is not shared with the Outgoing Line context.

Example Incoming Line DNIS =9234. Outgoing Line prefix = 9. This is a supported configuration.

To create an Outgoing Line navigate to Configuration > Telephony > Lines > Outgoing Line then Click “Add aNew Outgoing Line”. Enter a unique name for the new Outgoing Line. The name can match the dialpattern used for easier documentation of the configuration.

Choose the correct Trunk/Technology for this Outgoing Line. Choose the correct Interface Group ifapplicable. Click on the Dial String tab

10.0 Outgoing Lines


One of the most powerful and unique features in the ScopTEL IP PBX is theability to download the entire NPA‐NXX dial plan for any supportedArea Code and Prefix. This greatly simplifies the LCR (Least CostRouting) dial plan configuration for the server. Hours and possiblydays of configuration are reduced to seconds. However in thistutorial only a simple “Custom Dial String” option will be used.

After clicking on the “Dial String” tab choose “Custom Dial String” and thepage will automatically refresh.

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10.1 Outgoing Lines


Custom Dial Plan StringsX matches any digit from 0-9

Z matches any digit form 1-9

N matches any digit from 2-9

[1237-9] matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9)

. wildcard, matches one or more characters

! wildcard, matches zero or more characters immediately

Examples

NXXXXXX matches a normal 7 digit telephone number

1NXXNXXXXXX matches an area code and phone number preceded by a one

9011. matches any string of at least five characters that starts with 9011, but it does not match the four-character string 9011 itself.

# matches a single # key press

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10.2 Outgoing Lines


Dial String = Matching Pattern. Access Code = Outgoing DialPlan Prefix. This digit is always stripped so never passedto the physical interface.

Number of digit to strip = Number if leading digits strippedfrom the “Dial String”.

Prefix to add to Number = The digit(s) prefixed to theoutgoing call.

Authentication (PIN) can be used to force user authenticationbefore call is placed.

Once all fields are completed click on the “Dial Options” tab

10.3 Outgoing Lines


Dial Options must be configured only if youwish to provide additional features suchas call recording options and audio hookinherit.

It is often useful to have a unique Music OnHold source for each Outgoing Line ifthe user places an outgoing call on hold.

Once these fields are configured click on theCaller ID tab.

10.4 Outgoing Lines


On physical interfaces that support custom ANI to be set onoutgoing calls it is useful to define a global Name andNumber for outgoing calls.

Fill in the custom name and number for outgoing calls here ifthe Outgoing Line > Trunk supports custom ANI

Note that FXO interfaces do not support custom ANI but inthis example the custom “CallerID Number” and “CallerName” are configured.

10.5 Outgoing Lines


The Class of Service Manager is used to create objects used to assign permissions or restrictions to Outgoing Lines, Incoming Lines, Extensions, Feature Codes, or Applications.

The Class of Service Manager can be found by navigating to Configuration>Telephony>Manager>Class of Service

To add a new Class of Service object from the manager click on the link to “Add a new Class”.

Class of Service objects also control permissions and restrictions which vary depending on whether or not a Hot Desk Extension, Agent Extension, orRoom (Hotel) Restriction Feature code has been invoked. For example an extension can have a Class of Service which restricts long distanceOutgoing Lines when no Hotdesk Extension is logged but if a valid Hotdesk Extension logs in then the Outgoing Lines access is allowed.

There is no limit on the number of Class of Service objects which can be created. Therefore many COS objects can be added to create granularsecurity rules which can easily be applied to Outgoing Lines, Incoming Lines, Extensions, Feature Codes, or Applications.

The Class of Service is one of the last objects to be built during a new installation because many pre‐requisites are required.

Before a new Extension can be added a COS must be built so that the COS can be assigned to the new extension.

Before a non default Feature Code can be created the matching module must be configured.

Before a Feature Code can be included in a COS the Feature Code must be configured.

Before an Application can be assigned to a COS the Application must exist. It is therefore more efficient to create any required Applicationsprior to adding any new COS objects so that the COS does not have to be edited more than once.

Before an Outgoing Line can be included in a COS object the Outgoing Line must already exist.

It is best practice to leave Incoming Lines>Options>Class of Service configured to the default “System Default” setting. This is because the PSTNinterface also has a “System Default” COS value and the Incoming Line COS and the Interface COS must use matching values else incoming callswill fail. Unique requirements could dictate non default settings.

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11.0 Class Of Service (COS)


Example 1 = basic (restricted Feature Codes)

Click Add a new Class

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Give the COS a name (in this examplethe name is “basic”

Optional: fill in the description

Click on the Services tab to configurethe feature codes

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Click on the Select button to open a newwindow showing the available FeatureCode list.



In this example Call Recording, Channel Spy, Phrase Management, Do Not Disturb(DND), Call Forwarding options have not been allowed in the column to theright. Therefore if this COS is assigned to an extension that extension will notbe able to access those excluded features.

From the column on the left showing the available feature codes highlight eachfeature code required using a mouse and then click >> to assign those codes tothe column on the right. Click on the “OK” button to close this window.Feature Codes listed in the right column will be added to the new COS once thenew COS is saved.

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In this tutorial no Applications have been built.

If any real world Applications have been built the Selectbutton could be clicked to add access to anyApplications using the same method used to addFeature Codes.

Click on the “Local Extensions” tab to define access rulesfor Local Extensions.



In this tutorial “Enable All Local Extensions” will be checkedsince this COS will allow any Local Extension to be calledfrom any Extension with this COS assigned.

Local Extensions can be put into groups using COS objects. Thisis done by building multiple COS objects each with adifferent list of allowed Extensions and assigning thoseCOS objects to each Extension in the desired group. Thesame method is used to Select allowed Extensions as wasused to assign access to Feature Codes.

Click on the “Outgoing Lines” tab when finished configuringaccess to Local Extensions.



In this tutorial “Enable All Local Extensions” will be checked since thisCOS will allow any Local Extension to be called from any Extensionwith this COS assigned.

If multiple Outgoing Lines did exist then the “Select” button could beused to allow or disallow access to those lines.

Outgoing Line Examples (using outgoing PBX prefix=9):

9XXXXXXXXXX (10 digit local number)

91XXXXXXXXXX (11 digit North American Long Distance code)

To disallow the Long Distance Outgoing Line 91XXXXXXXXXX do notinclude this Outgoing Line in this COS.

Click on the “Miscellaneous” tab when finished assigning OutgoingLines.



In this tutorial separate COS values examples for logged AgentExtensions and or Hotdesk Extensions will not be given.

However if either of these options are checked then a COS touse when a logged Agent Extension or Hotdesk Extensionis not logged versus logged can be defined. Once this isdefined in this COS object then this COS object can beassigned to any Extension(s).

When this tab is configured click on the “Add” button to finishcreating this “basic” COS object.



Here is the finished COS object = basic

Additional COS objects should be built to provide moreadvanced permissive features access for moreadvanced users.

Each COS object requires a unique name.



SIP Phones are SIP User Agents.

For security, SIP User Agents must register to the SIP Registrar via username and password authentication.It is typical for the SIP protocol ports to be open or forwarded to the ScopTEL server if a third party Firewall is implemented.When the SIP ports are exposed on the Firewall it is common for hackers to attempt brute force attacks on the server. Such attacks systematically

request authentication using common dial plan Extensions and trivial passwords.Examples of such brute force attacks:

Extension range 100‐3000Systematic Password attempts using passwords 1000‐3000Systematic Password attempts using passwords 0000, 1234, 1111, 4321, 123456, 7654321

Therefore if a secure password policy is used it will prevent the overall majority of hackers from registering a SIP Extension or SIP Trunk with theserver for fraudulent purposes.

Examples of secure SIP password policyMinimum password length of 8 alpha numeric characters.No Dictionary wordsMinimum 2 Upper Case characters usedMinimum 2 numerals usedPasswords should be unique for each extension

The same policy enforcement should be in effect when configuring Voicemail Passwords except Voicemail Passwords cannot contain Alpha charactersand must be numeric.

A poorly implemented Voicemail Password Policy can allow a hacker access to thru dial capabilities from a mailbox configured to allow outdialcapabilities. Therefore Voicemail Passwords must be strict regardless of inconvenience caused to end users.Voicemail Password should never match the extension number.

Example: Extension 100, Voicemail Password 100Voicemail Password should never be trivial

Examples: 0000, 1234, 1111, 4321, 123456, 7654321

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12.0 Security


To set a Global Password Security Policy navigate toConfiguration > Telephony > Configuration > Security

The SIP and IAX2 Password Policy is set independently of theGlobal Voicemail Password Policy.

If the Options to automatically fix invalid password?[ ] is checkedthen non‐compliant passwords will be made compliantafter a commit (recommended). Here are therecommended Settings:

12.1 Security


It is common for SIP Extensions to exist for Remote Extensions (Nomadic users). It is highlyrecommended that the server be protected from malicious attacks by enabling the Firewall.

Configuration>Network>Firewall>General>Server Type

Server type is default with “No Firewall”. Firewall types are “Single System,Gateway/Firewall”

If only one Network Interface exists then only “Single System” or “No Firewall” is possible. Iftwo Network Interfaces exist then the server can be configured as a “Gateway/Firewall”which will enable outgoing NAT (Network Address Translation) and Firewall theconfigured WAN Interface.

In this screenshot the “Server Type” is configured as a “Single System” (Firewall is enabled). It is alsopossible to set the “Server Type” and “Inbound Services (Permit)” options using theConfiguration Wizard.

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12.2 Security


In this example the Firewall Configuration Wizard will beused to set the recommended Firewall Configurations.

From Configuration > Network > Firewall > General

Click on the “Configuration Wizard” button

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12.3 Security


In this example the Firewall Configuration Wizard will be used to set the recommended Firewall Configurations.

Note: In this example the server only has one Network Interface therefore “Gateway/Firewall” (NAT) is not possible.

From Configuration>Network>Firewall>General

Click on the “Configuration Wizard” buttonChoose the “Single System” optionClick “Next”

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12.4 Security


It is recommended to use thedefault checkboxes forInbound Services.

However if remote SIP Phonesmust be provisioned offsitethen it is possible to checkthe box for “TFTP Server” toallow the remote SIPPhone(s) to download theirrequired configuration files.Click “Next” when done

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12.5 Security


If the configuration is acceptable thenclick on the “Apply Changes?”checkbox and then click on“Finish”

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12.6 Security


From the Summary page click on “Close”

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12.7 Security


From Configuration > Network > General

Click on “Edit Services”

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12.8 Security


Check the “Firewall” option

Click on “Apply Change”

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12.9 Security


Click “Commit” to write the changes to the database.

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12.10 Security


Click on “Start Service”

Note that any future changes to the Firewallconfiguration will require another Commitand a Firewall Service Restart

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12.11 Security


Typical Extensions types are:

SIP Extension (IP Extension using the SIP protocol)

IAX2 Extension (IP Extension using the IAX2 protocol)

Zap Extension (analog FXS extension using Sangoma or Digium cards. Sangoma and Digium cards should not co‐exist in the same server)

Voicemail Extension (Voicemail box only)

Hotdesk Extension

A Hotdesk Extension is an Extension that logs into a physical Extension using the Hotdesk Feature Code, HotDesk Extension numberand required password.

By logging into a physical Extension the HotDesk Extension can make and receive calls from any extension which allows the HotDeskFeature Code in its assigned Class of Service. Caller ID incoming and outgoing will be automatically manipulated to displayHotDesk user information.

Virtual Extension

A Virtual Extension is a very advanced Extension type which allows a user to login to the ScopTEL GUI and use the Realtime Monitorand customize Call Detail Reports and other types of reports.

Advanced options can be configured to ring multiple destinations and automatically forward copies of voicemail messages to multipleextensions

User Options for Virtual Extensions include Follow Me, Camp‐On, Personal IVR destinations

Custom Forwarding Rules can be defined for:

Call Forward Immediate

Call Forward Busy

Call Forward No Answer

Call Forward Unavailable (forward when physical extension is offline)

It is possible to Immediate Forward a Virtual Extension to make an Application available within an IVR context for inboundPSTN callers.

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13.0 Extensions


Ring Group Extension

A Ring Group Extension automatically Immediately Forward it’s calls to configured Follow Me destinations

Advanced options can be configured to ring multiple destinations and automatically forward copies of voicemail messages to multipleextensions

User Options for Virtual Extensions include Follow Me, Camp‐On, Personal IVR destinations

Custom Forwarding Rules can be defined for:

Call Forward Immediate

Call Forward Busy

Call Forward No Answer

Call Forward Unavailable (forward when physical extension is offline)

It is possible to Immediate Forward a Virtual Extension to make an Application available within an IVR context for inbound PSTNcallers.

Shared Extension

A Shared Extension can be configured so that multiple extensions can ring when the pilot DN is dialed but depending on the busy status of theextension(s) one or more extensions can ring but the busy extension will not ring

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13.1 Extensions


To create a SIP Extension navigate to Configuration > Telephony >Extensions

Click on “Add a New Phone”

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13.2 Extensions


Choose “SIP” from the list of availableExtension types from the drop downmenu

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13.3 Extensions


The Username should match the numeric value of this Extensionnumber

Since the Security Policy enforces a strict SIP/IAX2 PasswordPolicy the first pre‐requisite is to enter a compliant alphanumeric password into the Authentication > Passwordtext box. Click on the General tab once the Authenticationtext is entered

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13.4 Extensions


The tenant is the default tenantThe type is IP to match the SIP end pointThe Extension number matches the

Authentication UsernameHardware ID is descriptive and does not

have to be specifiedClass of Service is mandatory and

determines what permissions areallowed

for this Extension.Full Name is used for the Voicemail

Directory and default CLIDpresentation

Description is not mandatoryClick on the Voicemail tab to configure

Voicemail options

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13.5 Extensions


Click on “Enable Voicemail” to enable Voicemailoptions

Since this is the first extension to be configured onthe server it will be the Global Operator for thistenant therefore:

“Act as an Operator” is checked

“Use User defined CallForward” will ensurethat the User Options Call Forwardingrules will be enforced.

The “Act as an Operator” option can beselected for additional Extensions if theyshould also ring when a caller dials 0.However it is typical for a system tohave only one Operator therefore thisoption should not be checked for otherextensions.

The Voicemail Password must comply to with theVoicemail Security Policy

The screenshot shows typical Voicemail settings

Many advanced configurations are possible from thispage including Voicemail to Email

When finished click on the “Phone Options” tab

13.6 Extensions


Host Mode should be left default and the IP addressfield should be ignored because this is an advancedfield used for problematic Remote Extensionsbehind a NAT Router

If the SIP device is to be used on the LAN then the“Phone behind NAT” option should not be checked.

If the SIP device is to be used as a Remote Extensionlocated behind a NAT router then the “Phonebehind NAT” option should be checked. Checkingthis option is normally sufficient to ensure that theRemote Extension can register with the server andtwo way speech paths are possible (assuming thatthe Firewall is and global NAT options areconfigured correctly)

Qualify is default and allows the server to monitorthe Extension for Registration status and packetlatency. But not all SIP devices support SIP Qualifyso this might have to be unchecked.

DTMF mode is normally RFC 2833Only CODEC’s supported by the SIP end point shouldbe checked

13.7 Extensions


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13.8 Extensions

Incoming/Outgoing Call Limit can restrict thenumber of simultaneous calls supported bythis Extension (default 8)

“SIP Alert (Auto Answer/Distinctive Ring)” is usedto configure this SIP end point to receivean internal page if the SIP end point is asupported device.


13.9 Extensions

In this example a Polycom SIP phone will beconfigured with supported CODEC’s andinternal paging requirements

Enable ‘SIP Alert‐Info pass‐through’ must bechecked

“Device” drop down menu selection mustselect Polycom

“Paging and Intercom” checkbox must bechecked

Click on the “Caller ID” tab when finished


All Caller ID fields can be modifiedDefault values will set the local and outgoing PSTN Caller ID

to match the configured Extension Number and NameUn‐checking either “Internal Call” or “External Call”

checkboxes will allow the Caller ID configuration to bemodified

Note that “External Call” and “Emergency Call” Caller IDcannot be customized if the ITSP or PSTN provider’strunks do not allow the Caller ID (ANI) to be re‐written

It is highly recommended that the “External Call“ and“Emergency Call” be modified to show either thepublished “BTN” of the customer or “DID” of the user.Failure to modify the defaults will result in only theName and Extension number appearing on anyoutgoing external calls.

Click on the “User Options” tab when finished modifications

13.10 Extensions


User Options define call forwarding rules,language, Music On Hold source filedirectory, default ring time, CallRecording options, Fax Detection

Enabling any advanced options such as“Follow Me”, “Personal IVR”, “Camp‐On”, “E911 Location” will add new tabsand options to this extension’s GUIinterface and allow additionalconfigurations

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13.11 Extensions


A Call Forward No Answer to Voicemailrule is very typical in phone systemconfigurations. So from the “CallForward on No Answer” drop downmenu select “Voicemail” to set upthe default rule to this Extensions’Voicemail box

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13.12 Extensions


Call Monitor option “On Demand CallRecording” allows a phone with a validClass of Service to record phone callsusing the Call Recording Feature Code

Call Monitor options can be configured to forcethe recording of all calls made to andfrom this extension using the “Force CallRecording” option

Click on the “Web Authentication” tab tocontinue

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13.13 Extensions


The “Web Authentication” option allowsthe owner of an Extension to loginto the ScopTEL GUI and accessseveral unique features

To access those features a unique loginis created by checking the “EnableUser Web GUI” and assigning aunique Username and Passwordfor this Extension

The user logs into the same IP addressand management port as theadministrator but uses this login toaccess their personal GUI login

Click on the “Security” tab when finishedwith this configuration

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13.14 Extensions


A password can be enforced when another extension or PSTNchannel attempts to call this extension. If the password is notentered correctly then the Extension cannot be called.

This setting is optional and rarely used

Click “Add” when finished to complete adding this extension to theserver

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13.15 Extensions


Incoming Lines must be created to Route incomingcalls to required destinations

From Configuration > Telephony > Lines > IncomingLines

Click on “Add a new Incoming Line”

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14.0 Incoming Lines


Incoming Lines types are typically:

“Extension (DNIS)” which are dialed numbers incoming on SIP/IAX2 or PRI trunks.

“Block” (a sequential list of DNIS numbers)

“Port (TDM)” which are analog FXO ports supported by Sangoma or Digium cards

There will be two examples. Example 1 will define incoming lines on analog FXO ports. Example 2will define incoming lines on a SIP trunk.

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14.1 Incoming Lines


Analog FXO ports are port based trunks

The Incoming Line is used to define a destination for one or more FXO ports.

Additional options allow to define custom schedules, Caller ID Prefixes, Music On Hold sources, CallRecording Options

From the drop down menu list select “Port (TDM)”

Click on the “Select” button to open a new window and add the required ports to the Incoming Line

Click on “OK” when the required ports have been added

Click on the “Destination” tab next6464

14.2 Incoming Lines


DNIS rules apply to T1/E1 trunks with ISDN signaling and many ITSP SIPtrunks route the same method

The Telco forwards the dialed number to the physical interface so that thisinformation can be used to route many dialed numbers to manydestinations using the same physical interface. This is a very flexibleand highly efficient method to route multiple numbers to variousdestinations using a single trunk

There is no fixed length requirement for the DNIS digits

There is no restriction preventing any Outgoing Line prefixes from matchingthe leading digit of any incoming DNIS digits

This is Example 2

The Options tab must be configured to use the required SIP trunk inthe “Trunk” selection menu

All other tabs are common to both port based trunks and DNIS basedtrunks

14.3 Incoming Lines


Set Destination #1 to the first requireddestination

In this example the Incoming Line will ringDestination Extension 100

Choose the “Extension(s)” option from themenu drop down list

14.4 Incoming Lines


It is typical for the forwarding rules defined inthe Extensions’ User Options to berequired therefore the “Use User‐definedCallForward” checkbox needs to bechecked else additional destinations mustbe configured on this Incoming Line

Click on the “Options” tab when finishedconfiguring the Destination

14.5 Incoming Lines


Choose an already configured Trunk from theTrunk drop down selection. In thisexample the SIP trunk = 5555551234 isused.

The “Options” tab is also used to set sharedIncoming Line fax detection, language,and the Music On Hold source amongother options

Click on the “Caller ID“ tab after reviewing

14.6 Incoming Lines


The Caller ID tab is used to add additional Caller IDprefixes or source routing to Incoming Lines

In this example since the Outgoing Line that wasconfigured requires a leading prefix of 9 a 9will be added to the Incoming Caller ID prefixfor each incoming call number. This adds the 9to any incoming caller ID history featureprovided by supporting SIP end points so thatthe user does not have to edit the call historyto add the 9 prefix to the existing call history

In this example the “Company ABC:” prefix is addedto the incoming Caller ID name so that areceptionist/operator can easily distinguishinbound calls for each customer and answerthe phone with the correct greeting response

Click on the “Add” button when finished adding anIncoming Line

14.7 Incoming Lines


The APS (Automatic Provisioning Service) isused to create the required configurationfiles needed for many SIP end points towork correctly

The APS assigns SIP usernames and passwords,network options, time settings, QoSsettings, dial plan options, firmwareupgrade policies, soft key programming,DSS/BLF programming, security settings,DTMF modes, LDAP settings,

Templates can be configured to simplify tediousconfiguration settings for as manysupported SIP end points as required

From Configuration > Telephony >Provisioning click on “Add a newProvisioning System”

15.0 Automatic Provisioning Service


A template should be configured first tosimplify the creation of many supportedSIP end points

In this example a template will be created for aPolycom IP550 SIP phone

The default tenant is selected

The phone model is selected from the dropdown menu of supported end points

The “Create Template” checkbox is checked

The template is given a name

Click on the “Firmware” tab when theseoptions are configured



The correct firmware stream must be selected

If Polycom firmware version 3.0.4 is applicable to the hardwarethen choose the “Use new provisioning” option

When all options are configured click on the “Lines” tab

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It is highly advantageous to “Customize number of Lines/Soft Keys”

The recommended “Number of Line” is 1

The maximum “Number of Soft Key” for this device is 4

When these options are configured a new “Soft Key” tab will appear inthe GUI

Click on the new “Soft Key” tab when finished configuring theseoptions



For each supported key choose an option:

“None” leaves the key empty

“Feature” allows to select a pre‐configured Feature Code

“Speed Dial” allows to configure an internal or external speed dial key. If the“Buddy Watch” checkbox is checked, then this enables BLF (Busy Lam pField) on this key

In this example all three Soft Key keys are configured as feature codes

Click on the “Servers” tab when finished configuring these options

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The SIP Proxy address must be configured. The address must be routable on the network.

The Outbound Proxy address must be configured. The address must be routable on thenetwork.

In this example the eth0 address is used. This address will vary per installation and willprobably not be the same as in this example.

The IP address can use the optional “IP/DNS Mapping” feature if a FQDN (Fully QualifiedDomain Name” is preferred but a DNS A Record must exist on any DNS server(s) toresolve this FQDN. It is advantageous to use a FQDN address so that the SIP phone cancommunicate with the server either by it’s local or public IP address. This makes the SIPphone very portable.

When these options are configured then click on the “Network” tab



The “Line Label” option can be used to displayeither the provisioned extensionnumber or name on the idle LCD displayof the Polycom IP550

The “Dial Plan” text field can be customized tosupport “En Bloc” signaling as per thePolycom SoundPoint Admin Guideapplicable to the correct firmwarerelease

Click on the “Date and Time” tab whenfinished configuring these options



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The default “Date and Time” settings areconfigured for North American DaylightSavings Time Rules

These settings can be modified as per thePolycom SoundPoint Admin Guideapplicable to the correct firmwarerelease

Click on the “User Preferences” tab whenfinished configuring these options


“User Preferences” can be edited to suit therequirements of the users

The default settings are acceptable so afterreviewing this page click on the“Audio/Ring Tone” tab



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A majority of users prefer to disable “Local UIsound effect”

A majority of users prefer to select PersistentVolume settings

A majority of users prefer to disable Stuttered DialTone for Message Waiting Indication. Thiscan be disabled since the Polycom IP550supports a MWI light

Click on the “Security” tab when finishedconfiguring these options


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It is advised to uncheck the “Enable Web Server” option toprevent users from logging into their phones and makingchanges or harvesting configuration settings formalicious intent

When finished configuring these settings click on the “Add”button.


Configuration files are created and most SIPend points including Polycom downloadtheir required configuration files withnames based on their MAC address

To create a SIP end point specific configurationfile for a supported end point click on“Add a new Provisioning System”



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Choose the correct model number from the “Phone Model”drop down menu

Choose the “polycomip550” template that was just created fromthe “Phone Template” drop down menu

Fill in the “MAC Address” text box with the MAC address of thedevice

When finished click on the “Lines” tab


From the “Key/Line 1” drop down menu select therequired Extension from the available list. It isonly necessary to add the extension to the Line 1assignment since the Polycom firmware will allow8 calls to be handled on this key using the LCDdisplay.

Click on the Micro‐Browser tab when finishedconfiguring this option

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The Micro‐Browser feature for Polycom phones is a featureunique to ScopTEL. This feature pushes phone statusand ACD Wallboard Features directly to the idle LCDscreen of each supported Polycom SIP end point.Supported devices include the IP450, IP550, IP650,IP670, IP500, IP600, IP601.

To enable this feature check the “Use Internal PBX services(Idle/Main)” checkbox

From the “Extension” drop down menu select the requiredExtension number to monitor the status of thisExtension

When finished configuring these options click on the “Add”button

Polycom phones must have their default boot optionschanged in order to download provisioning files

The default method is FTP. ScopTEL does not supportprovisioning via FTP.

From the Polycom Administrative Menu, Network Optionsedit the Server Menu option to use TFTP or HTTP andreboot the device. 8484



Add as many COS objects as required, Add as many Extensions and APS files as required, Add asmany Incoming Lines and Outgoing Lines as required

More advanced features such as schedules and multiple Music On Hold sources can be added byreferring to their supporting documentation

From Configuration > Telephony >GeneralWhen all is configured then all changes must be Committed to write them to the live databases.

Click the “Commit” buttonThe Telephony and Zaptel services must be restarted to properly load the required Zaptel/Dahdi

settings into memory. To restart those services first navigate to the “General” tab…The correct order to reset services is:

Stop the “Telephony Server”Restart the “Analog/Digital Modules (Zaptel/Wanpipe) ServiceStart the “Telephony Server”

16. Commit and Services Restart


Congratulations !

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ScopServ - Basic Installation Hierarchy for Telephony Server New Design v04_2.pdf

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