Category: How To Guides:

A comprehensive list of how to guides for asterisk and voip gear.

Polycom Electronic Hook Switch Guide

This .e4 report entry shows phone administrators and end users how to use a Plantronics wireless headset and electronic hookswitch (EHS) with Polycom SoundPoint IP  and VVX Business media phones.

Some Plantronics headsets enable you to answer and complete phone calls using the built in controls located on the headset. This feature is called electronic hookswitch (EHS). To use it, your headset must include a digital serial control interface in addition to the standard analog headset jack (RJ-9) interface. The serial control interface requires an APP-51 which you can purchased from e4. All Plantronics Headsets that support EHS come with a base unit that connects to your phone. The base unit and headset will connect wirelessly.This enables you to answer the phone when away from your desk by pressing the button on your headset. All of the Plantronics headsets in the guide have a wireless range of over 300′ from the phone.

Supported Phones:

As of 3/14/2013 the following Polycom SoundPoint IP phone support EHS. IP 320, IP 321, IP 330, IP 331, IP 335, IP 430, IP 450, IP 550, IP 560, IP 650, IP 670.

As of 3/14/2013 the following Polycom VVX Business media phones support EHS. VVX 300, VVX 310, VVX 400, VVX 410, VVX 500, VVX 600, VVX 1500.

Supported Plantronics Headsets:

This guide will focus on the PlantronicsSavi 700 series, CS500 Series and Plantronics MDA200 device which enables the management both Polycom desk phones and PC calls.

Please note: Legacy Plantronics Savi Office models such as  the WO100, WO200, WO300, WO350 and Plantronics CS models 50, 55, 60, 70N, 261N, and 351N  support EHS with Polycom phones using the APP-50 EHS cable and APP-5 cables respectively.
Plantronics Savi 700 Series:

Plantronics CS500 Series:

  • CS540 Convertible P/N 84693-01
  • CS530 Over-the-ear P/N 86305-01
  • CS520 Over-the-head (binaural) P/N 84692-01
  • CS510 Over-the-head (monaural) P/N 84691-01

.e4 also carries the Microsoft Lync enhanced versions of the these headsets for those using the MS UC platform.

The following image set demonstrates the proper setup for using Plantronics headsets with electronics hookswitch support and you Polycom telephones.

All headsets listed in this article are available through .e4. Please contact Daniel Bonham our resident headset expert at 877-434-8647 x102 to obtain special pricing and advice on which Plantronics model is right for you.

 

SMS Notification with Switchvox

Ever wonder if you could text messages from your Switchvox? If you’ve answered yes you will be happy to know that you can…  Setup is simple and the cost is just a penny.

Objective: Send a text message to one or more people by dialing an extension on a Switchvox phone system.

Here are a few scenarios for this feature…

  • Panic Button – Dial an extension that sends specific emergency instructions to a predetermined contact list.
  • Route Notification – Receive a text message when someone dials a specific number.
  • Call Volume Notification – Send an SMS message that notifies one or more people that call volume and customer service levels have been reached.

Using the API – In order to properly set up text message notification from Switchvox you will first need to understand how to formulate a get request using our API. Click here for additional calls such as CNAM, LRN, and SMS MDR.

The following example assumes that you have an .e4 account in good standing with at least 1 DID.  If you don’t have an account you can signup here.

  1. Log in to your .e4SIP account – If you have an account you can use a simple login string to access the portal. https://username:password@portal.e4sip.com
  2. Access your unique API key.  When logged in your key will appear on-screen.

Sending an SMS is simple. Our example will allow you to send a text message directly from any browser’s address bar. Once you have your account and api key requests can be assembled in the format below.

Short Message Service (SMS)

https://api.e4sip.com/sms/[destination_number]/[caller_number]/[message]/[api_key]

RESTful GET Request

Property Description
[destination_number] Destination Number in NPANXXXXXX format
[caller_number] .e4 Phone Number (DID) in NPANXXXXXX format
[message] URL Encoded plain text SMS message – Max length 160 characters
[api_key] Your API Key

SMS is intended for a single destination number. Please use Group SMS to send a message to multiple destination numbers.

JSON Response

Property Description
Status REST Response
Message Detailed Response Message

Group Short Message Service (Group SMS)

https://api.e4sip.com/groupsms/[destination_number]-[dst_#2]-[dst_#3]/[caller_number]/[message]/[api_key]/

RESTful GET Request

Property Description
[destination_number] Multiple Destination Numbers in NPANXXXXXX format and separated by a dash
[caller_number] .e4 Phone Number (DID) in NPANXXXXXX format
[message] URL Encoded plain text SMS message – Max length 160 characters
[api_key] Your API Key

Group SMS is intended for use with multiple destination numbers. Please use SMS for a single destination number.

JSON Response

Property Description
Status REST Response
Message Detailed Response Message
Number Destination Number

Now for the Switchvox configuration…

1. Create an IVR- Tools > IVR Editor > Create IVR Menu

   2. Create IVR Action – Create Action > Action Type: Send Call Variables to URL.

   3. Copy Group SMS URL String from above and paste in to URL field.

*** DON’T FORGET THE TRAILING FORWARD SLASH – THE MESSAGE WILL FAIL WITHOUT IT.

Understanding the String…

https://api.e4sip.com/groupsms/

This section denotes which API call you are making – There are other calls that can be found here.  As mentioned, our example here is for group SMS but standard single recipient requests are made using a similar format.

/2318675309-5551234567/

In my example we are sending an SMS to 2318675309 and 5551234567.  You can add as many numbers as you would like while using our group SMS feature by separating each number with a hyphen as we have in the example.

/2319464162/

The 2319464162 number is our .e4SIP DID and a required parameter, you can only send an SMS message from numbers that are provisioned in your .e4 SIP account.  If someone replys to this request you will receieve an email with their message. This email will be delivered to the master account email in your .e4 SIP account.

/This is a group SMS test/

This is the section of your request that has the message you want to send. This message length is restricted to 160 characters.

/e532fdsdfss0955da548712faf956bdeb3/

This section is your unique API key. Don’t forget the trailing slash, it wont work without it. The one listed in our example is simply filler – You will require your own key to make this work.

From here there are many options.  In my example I set two additional actions for the IVR.  One that plays the default thank you sound and another that ends the call with the hang up action.

  4. Create Extension: Setup > Manage > Create Extension: Extension Type IVR

   5. Issue the extension number that you will dial to initiate the SMS notification.

   6: Test: Dial the extension number that you have created and Viola!

Contact .e4 today for more information on this and other exciting .e4 Switchvox related features, hardware and services.

 

 

 

 

 

 

 

 

 

How to: Aastra Web Recovery Mode

UNBEATABLE PRICING IN OUR STORE! CLICK HERE TO CHECK IT OUT!

The Aastra web recovery mode allows your phone administrator to restore a phone to it’s factory state in the unlikely event that your Aastra SIP phone’s default firmware becomes corrupt.

This issue is generally reported after the device was interrupted while downloading the required *.st binary package or after switching from PoE to AC.

When your Aastra SIP phone is in web recovery mode the phone will power up and the LCD will be blank. At this time the phone is unusable. To start the recovery process please refer to the steps below. In our example we will be using PBXact. You can also utilize the steps listed by pointing the phone at your own TFTP server.

Aastra Web Recovery Mode Step by Step:

  1. Reboot your Aastra SIP Phone
  2. As it  reboots quickly press and hold 1 + # untill the phone’s IP address is displayed.
  3. From a computer on the same network as the Aastra phone, visit the IP displayed
  4. You will see many fields on this page- only 2 are used:
    1. Filename:
    2. TFTP Server IP:
  5. In the Filename field, enter the name of your phone’s model firmware file.  Aastra firmware file naming convention is a follows – See list below for the correct entry.
    1. Aastra Firmware files are named:
      9133i.st
      9112i.st
      480i.st
      480i Cordless.st
      480i CT.st
      51i.st
      53i.st
      55i.st
      57i.st
      57iCT.st
  6. In the TFTP Server IP field, enter the IP address of your PBXact or TFTP server .
  7. Click Download Firmware button
  8. The phone will then contact your TFTP server and download the file listed in the file name field. Please be sure to let the phone return to default before disconnecting the device.

Need help with you Aastra phone or Asterisk PBX? Not sure what phone system is right for your business?. Click Here to reach our expert team by chat or call us any time at 877.434.8647

Switchvox How To: internal Peering

If you have been working with Switchvox for any amount of time you know that
the system is jammed full of useful features that can be deployed with the click
of a mouse and a few keystrokes.  Over the years, as my Switchvox skill set
grew, I’ve noticed that you can create many workarounds simply by mashing
multiple features together and by trying things that you wouldn’t think will
work.

I read an interesting post on the new Switchvox forum over at
Digium.com  today.  The poster
asks the question, is there a way to alter caller ID on demand for internal
extension.  The answer is yes. Not sure exactly why you’d want to do this
but It is completely possible. For this we’ll make use of a small work- around
that I figured out some time ago that can be used for so many wonderful things.

Peering Switchvox to itself: This workaround was originally
discovered when I was not going to get a deal based on the system’s inability to
do six digit extensions. (This is now a function of SWVX) this workaround saved
the day, and we’ve won the customer and are rolling roughly four systems a month
to this very same customer.

Ultimately,  this workaround is a testament to versatility of Switchvox.

Creating the Peer:

For this we’ll use the exact same methodology as peering two Switchvox
systems together but in this example we have only one Switchvox server. Today
I’m using our
Switchvox AA60 SMB.

Logged in as admin, navigate to: System Setup > VOIP Providers.

You’ll need to create two new IAX Providers

Select IAX Provider from the Add New: Pull Down Menu then press go.

Create a new IAX Peer with these specs:

The Create a Second IAX Peer with these settings:

Next Navigate to: System Setup > Outgoing Calls > Add New Outgoing
Rule.

Create Rule Like this-  Can be tailored to match your Extension
numbering.

Create Caller ID Change:

Next Navigate to: System Setup > Outgoing Calls > Add New
Caller ID Rule

In this Example the calling extension will first dial their Extension number
then the Extension number that they want to call with the altered caller ID
state.

caveat – Be sure that your phone’s dial plan can support the additional
numbers when calling extensions. adding x’s to your dialplan may result in
needing to press send or dial on your phone.

Polycom Example

[0-8]xxxxxx|911|9411|9611|9011xxx.T|91xxxxxxxxxx|9[2-9]xxxxxx|*xx.T

Here’s what it looks like when EXT 103 Calls EXT 100 (My New Cisco
SPA508G)

Hopefully you find this useful… After tinkering with this for a while I
found that there are quite a few things you can do with this – A few other
examples of what I have done with this are creating “Meet Me” conference rooms
with pin numbers on Switchvox Free Edition and also creating passwords for
getting to our paid after hours support product.

Enjoy,

MW

Digium Telephony Cards

Originally created by Alexander Graham Bell in the late 1800’s, and having not changed significantly since, analog telephony uses a pair of copper wires to both send and receive sound. Each pair carries one phone call at a time. At one end of the line is the telephone company’s switching office and at the other end is the phone. In telco jargon, the phone company’s gear is referred to as Foreign Exchange Office or FXO and the phone set is Foreign Exchange Station or FXS.

How do you know to answer the phone? Because it rings when voltage is supplied by the phone company. And, when you take the phone off-hook, a voltage change signals the phone company that you’ve answered, beginning the call. Using the same simple signaling system Asterisk can connect to the telephone company through FXO ports, also known as line ports. Asterisk can also connect with and send calls to analog telephones and other analog devices through FXS or station ports.

Each analog line you want to connect with Asterisk requires its own analog line port (FXO port). Each analog phone you want to hook up requires its own station port (FXS port). Asterisk connects to analog lines and stations using interface cards which are installed in the system’s PCI or PCI-Express slots. Digium offers analog interface cards that range from one to twenty four ports. Because analog lines are prone to echo, an optional hardware echo cancellation module is available for all card models.

Asterisk Tip

When setting up analog devices in Asterisk’s DAHDi configuration file, you must specify that FXO devices use FXS signaling and FXS devices use FXO signaling. This is not an bug or a flaw in the documentation. Really.

; Line (FXO) Ports 1 and 2
signaling = fxs_ks
channel = 1-2

; Station (FXS) Port 3 – Fax Machine
signaling = fxo_ks
channel = 3

ISDN BRI Lines

ISDN-BRI circuits are a type of digital line designed for small businesses and home offices and are found almost exclusively outside of North America. BRI is much like ISDN-PRI, only supporting a reduced number of channels. Each BRI line can carry two simultaneous calls. Digium offers a 4-port PCI-based BRI interface card with on-board echo cancellation that supports up to 8 calls.

Card Configurations

The type of interface you choose to install is usually determined by the size of your business and the type of telephony connections available. Analog connections are generally a more economical choice for small businesses. If your system requires more than about 8 analog lines you should check with your carrier to see if a digital connection would save you money.

If you are using Asterisk to build a VoIP gateway for a legacy PBX system you may have several options for connecting. Larger legacy systems frequently include T1 or E1 digital ports while smaller PBXs and key systems have analog line or digital BRI ports. To connect Asterisk with the legacy system using T1 or E1 digital lines, you will need a digital card and a “cross-over” cable; digital BRI connections use a “straight-through” cable. For analog connections you will need a card with FXS interfaces (station ports) to connect with the FXO ports (line ports) on the PBX. In either case Asterisk will appear as the phone company to the legacy system.

Asterisk systems frequently include a mix of analog and digital interfaces. Digium’s analog cards are modular and can include both line ports (ports that connect to lines from the telephone company) and station ports (ports to which you connect an analog phone). To easily determine the right set of cards for your deployment, check out the card selector utility.

We hope this helps you understand the basics of PSTN telephony with Asterisk. For detailed information please check out the tutorials and information on Asterisk.org and Digium.com.

Digital Lines

Digital lines or “trunks” allow for multiple calls across a single circuit. Telephone companies in the United States offer digital service over T1 lines which can handle up to 24 simultaneous calls. Europe and much of the rest of the world uses the E1 standard which allows for 30 simultaneous calls. T1 and E1 lines use a technique known as “time division multiplexing” to send multiple calls across the circuit. The total capacity of the line is divided into 24 (T1) or 30 (E1) “time-slots” or “channels”, each of which can handle a single voice call or 64 Kbps of data.

Unlike analog connections, digital lines have several options for call signaling (the process of setting up and tearing down calls). Older systems tend to use “in-band” signaling, which sends messages as tones or special patterns in the digital data stream for each call. Newer systems often use “out-of-band” signaling methods which allocate a special signaling channel (one of the time-slots on the circuit) to handle all call setup and tear down messages and to exchange other call-related data. Most digital lines with out-of-band signaling use the ISDN-PRI standard. Another less common implementation uses the SS7 standard.

Asterisk connects to T1 and E1 voice lines using PCI or PCI Express gateway interface cards. Digium’s line of digital cards include one port (24/30 calls), two port (48/60 calls) and four port (96/120 calls) models. As with the analog interface cards, echo cancellation modules are available for each model.

For more information on Digium Cards and other Digium hardware like Digium Phones and gateways be sure to check our the Digium Section of our store.

Thanks for stopping by!

Aastra 480i Factory Reset

Boot Server Setup

480i and/or CT

  1. Press the Options key (Aastra A logo) key below the redish key.
  2. Scroll down to 9. Network and press the ‘Show’ softkey.
  3. Enter the password: 22222 and press the ‘Enter’ softkey.
  4. Scroll down to 7.TFTP Server and press the ‘Show’ softkey.
  5. Scroll to 2. Alternate TFTP and press the ‘Show’ softkey.
  6. Set the IP address to <IP ADDRESS OF TFTP SERVER> and press the ‘Done’ softkey
  7. Scroll to 3. Select TFTP and press the ‘Show’ softkey.
  8. Press the ‘Change’ softkey until the ‘Select TFTP’ option is set to ‘Alternate’ and press the ‘Done’ softkey
  9. Keep pressing the ‘Done’ Softkey until you are presented with the ‘Are you sure you wish to restart the phone?’ and press the ‘Restart’ softkey.

Via the Web

  1. Find your phone’s IP Address
  2. Enter the phoneï¾’s IP Address in to a web browser on a computer on the same subnet as the phone.
  3. User Name = admin, Password = 22222
  4. Select Reset from the menu on the left
  5. Select Restore
  6. The phone will reboot, revert to factory defaults, and then attempt to download it’s configuration file

Via the phone

  1. Press Options
  2. Press Phone Status
  3. Press Restore Defaults
  4. Enter the password 22222
  5. Press All Defaults
  6. The phone will reboot and download software and configuration files from provisioning server.

Adding a Linksys SPAxxx to Switchvox – WEB GUI Method

First find the phones IP address-
  1. Press the menu key (looks like a piece of paper folded at the right corner – near the hand key)
  2. Press 9 for “network”
Once you have the IP address of the phone, enter that IP in to your web browser. at this point you should see a web page that says Linksys SPA configuration in the title bar.
  1. Click the link at the top that says “admin login”
  2. Click the link that says “EXT1”

In our example the configured extension is 120 and the password is 120 – Our Switchvox server resides on the local LAN @ 192.168.1.22
Dont for get to click “submit all changes”
The information listed above will register your phone to Switchvox however there will be a few more steps to fine tune the configuration.
Next:
  1. Select “advanced” on the top right
  2. Select EXT 1 Link (Again)
  3. Find the are labled  “Dial Plan”
  4. Enter (xxx|9[2-9]xxxxxx|91xxx[2-9]xxxxxxS0|9xxxxxxxxxxxx.)
  5. Hit Submit
This Dial plan Example assumes you have 3 digit extension – 4 and 5 digit examples are listed below – There are a few things that can be done to make this dial plan more effective- the rule listed below will work for users dial 91+10 digits for long distance and users dialing either 7 or 10 digits for local.
4 Digit Extensions:    (xxxx|9[2-9]xxxxxx|91xxx[2-9]xxxxxxS0|9xxxxxxxxxxxx.)
5 Digit Extensions:    (xxxxx|9[2-9]xxxxxx|91xxx[2-9]xxxxxxS0|9xxxxxxxxxxxx.)
Click “Submit all changes”
Voicemail setup:
  1. Navigate to the “phone” tab
  2. Find field named “Voice Mail Number:”
  3. Enter voicemail Extension Number (Usually 899 in Switchvox)
Click “Submit all changes”
Viola!
Follow Up –
Here is a dial plan that PREPENDS the 9 for calling from the redial list…
(xxx|9[2-9]xxxxxx|91xxx[2-9]xxxxxxS0|9xxxxxxxxxx.|<:9>xxxxxxxxxx)

Aastra Mega Conf circa 2006

####################################################################################################################
# This file is designed as a guide to creating a <mac>.cfg file for your Aastra IP Phones.
# All comment, preceeded by a “#”, will be omitted from the config parser.
#
####################################################################################################################
#
# .e4
# Updated: 25/7/06
# Firmware: 1.4.0.1048
#
####################################################################################################################
# Security options
####################################################################################################################
#
options password enabled: 0 # 0 = false (default), 1 = true
admin password: 22222 # up to 63 alpha numeric chars
user password: # up to 63 alpha numeric chars (blank by default)
options password enabled: 0 # allows password protection to the “options” menu.
# # Password is the admin password and must be
# # entered in 3 attempts.
# # 0 = false (default), 1 = true
#
telnet enabled: Aastra Telecom Inc # 0 = disabled (default), Aastra Telecom Inc = enabled
web interface enabled: 1 # 0 = false, 1 = true (default)
#
####################################################################################################################
# Misc Phone Side Options
####################################################################################################################
#
live dialpad: 0 # 0 = false (default), 1 = true
#
redial disabled: 0 # 0 = false (default), 1 = true
conference disabled: 0 # 0 = false (default), 1 = true
call transfer disabled: 0 # 0 = false (default), 1 = true
#
map redial key to: # map to a speeddial if a number is configured
map conf key to: # map to a speeddial if a number is configured
#
call forward disabled: 0 # Globally enable or disable call forwarding
# # option from the WEBUI and TUI
# # 0 = false (default), 1 = true
#
callers list disabled: 0 # 0 = false (default), 1 = true
missed calls indicator disabled: 0 # 0 = false (default), 1 = true
directory disabled: 0 # 0 = false (default), 1 = true
#
displayName1: # idle display name 1 (480i / 480iCT only)
displayName2: # idle dispaly name 2 (480i / 480iCT only)
#
stutter disabled: 1 # disable or enable stuttered dial tone when there is a message waiting.
# # 0 = false, 1 = true (default)
#
call waiting tone: 1 # enable or disable playing a tone when there is a call waiting while on
# # an active call.
# # 0 = false, 1 = true (default)
#
priority alerting enabled: 1 # 0 = false, 1 = true (default)
# # For Sylantro specific alerting tones see Pg 167 of the 1.4 Admin # # Guide.
#
language: 0 # 0 = English (default), 1 = French, 2 = Spanish, 3 = German,
# # 4 = Italian
#
suppress dtmf playback: 0 # enable or disable the playing of dtmf tones when a number is
# # dialed from a softkey or prgkey.
# # 0 = false (default), 1 = true
#
sip intercom type: 1 # Determines whether the IP phone or the server is responsible for
# # notifying the recipient that an Intercom call is being placed.
# # 1 = phone-side, 2 = server-side, 3 = off (default)
#
sip intercom prefix code: *96 # The prefix to add to the phone number for server-side outgoing
# # Intercom calls. (example: *96 is used by Sylantro)
#
sip intercom line: 1 # only used when Intercom type = phone-side
sip intercom mute mic: 1 # 0 = false, 1 = true (default)
sip allow auto answer: 1 # 0 = false (Intercom call is rejected), 1 = true (default)
#
audio mode: 1 # configure how the “handsfree” key operates.
# # 0 = speaker (default), 1 = headset, 2 = speaker/headset,
# # 3 = headset/speaker
#
directed call pickup: 1 # 0 = disabled (default), 1 = enabled
#
play a ring splash: 1 # Enables or disables the playing of a short “call waiting tone”
# # when there is an incoming call on the BLF monitored extension. If # # the host tone is idle, the tone plays a “ring splash”.
#
sip blf subscription period: 180 # range: 120 – 7200 seconds (default = 3600)
#
####################################################################################################################
# TIME SERVER SETTINGS
####################################################################################################################
#
time server disabled: 0 # 0=false, 1=true
time server1: 10.60.1.22
time server2:
time server3:
time zone name: GB-London # see below for all zone names
time zone code: GMT # See below for all zone codes
time zone minutes: 60 # manually adjust the time zone can be – or + value
time format: 1 # (0=12hr, 1=24hr)
date format: 4 # 0: WWW MMM DD, 1: DD-MMM-YY, 2: YYYY-MM-DD, 3: DD/MM/YYYY
# # 4: DD/MM/YY, 5: DD-MM-YY, 6: MM/DD/YY, 7: MMM DD
dst minutes: # manually adjust the daylight saving time using – or + integers
dst start relative date:
dst start month:
dst start day:
dst start week:
dst start hour:
dst end relative date:
dst end month:
dst end day:
dst end week:
dst end hour:
#
#The following are the names of the time zones and their corresponding time zone codes:
#
# AD-Andorra CET AG-Antigua AST AI-Anguilla AST AL-Tirane CET AN-Curacao AST
# AR-Buenos Aires ART AS-Pago Pago BST AT-Vienna CET AU-Lord Howe LHS AU-Tasmania EST
# AU-Melbourne EST AU-Sydney EST AU-Broken Hill CST AU-Brisbane EST AU-Lindeman EST
# AU-Adelaide CST AU-Darwin CST AU-Perth WST AW-Aruba AST BA-Sarajevo EET
# BB-Barbados AST BE-Brussels CET BG-Sofia EET BM-Bermuda AST BO-La Paz BOT
# BR-Noronha FNT BR-Belem BRT BR-Fortaleza BRT BR-Recife BRT BR-Araguaina BRS
# BR-Maceio BRT BR-Sao Paulo BRS BR-Cuiaba AMS BR-Porto Velho AMT BR-Boa Vista AMT
# BR-Manaus AMT BR-Eirunepe ACT BR-Rio Branco ACT BS-Nassau EST BY-Minsk EET
# BZ-Belize CST CA-Newfoundland NST CA-Atlantic AST CA-Eastern EST CA-Saskatchewan EST
# CA-Central CST CA-Mountain MST CA-Pacific PST CA-Yukon PST CH-Zurich CET
# CK-Rarotonga CKS CL-Santiago CLS CL-Easter EAS CN-China CST CO-Bogota COS
# CR-Costa Rica CST CU-Havana CST CY-Nicosia EES CZ-Prague CET DE-Berlin CET
# DK-Copenhagen CET DM-Dominica AST DO-Santo Domingo AST EE-Tallinn EET ES-Madrid CET
# ES-Canary WET FI-Helsinki EET FJ-Fiji NZT FK-Stanley FKS FO-Faeroe WET
# FR-Paris CET GB-London GMT GB-Belfast GMT GD-Grenada AST GF-Cayenne GFT
# GI-Gibraltar CET GP-Guadeloupe AST GR-Athens EET GS-South Georgia GST GT-Guatemala CST
# GU-Guam CST GY-Guyana GYT HK-Hong Kong HKS HN-Tegucigalpa CST HR-Zagreb CET
# HT-Port-au-Prince EST HU-Budapest CET IE-Dublin GMT IS-Reykjavik GMT IT-Rome CET
# JM-Jamaica EST JP-Tokyo JST KY-Cayman EST LC-St Lucia AST LI-Vaduz CET
# LT-Vilnius EET LU-Luxembourg CET LV-Riga EET MC-Monaco CET MD-Chisinau EET
# MK-Skopje CET MQ-Martinique AST MS-Montserrat AST MT-Malta CET MX-Mexico City CST
# MX-Cancun CST MX-Merida CST MX-Monterrey CST MX-Mazatlan MST MX-Chihuahua MST
# MX-Hermosillo MST MX-Tijuana PST NI-Managua CST NL-Amsterdam CET NO-Oslo CET
# NR-Nauru NRT NU-Niue NUT NZ-Auckland NZS NZ-Chatham CHA PA-Panama EST
# PE-Lima PES PL-Warsaw CET PR-Puerto Rico AST PT-Lisbon WET PT-Madeira WET
# PT-Azores AZO PY-Asuncion PYS RO-Bucharest EET RU-Kaliningrad EET RU-Moscow MSK
# RU-Samara SAM RU-Yekaterinburg YEK RU-Omsk OMS RU-Novosibirsk NOV RU-Krasnoyarsk KRA
# RU-Irkutsk IRK RU-Yakutsk YAK RU-Vladivostok VLA RU-Sakhalin SAK RU-Magadan MAG
# RU-Kamchatka PET RU-Anadyr ANA SE-Stockholm CET SG-Singapore SGT SI-Ljubljana CET
# SK-Bratislava CET SM-San Marino CET SR-Paramaribo SRT SV-El Salvador CST TR-Istanbul EET
# TT-Port of Spain AST TW-Taipei CST UA-Kiev EET US-Eastern EST US-Central CST
# US-Mountain MST US-Pacific PST US-Alaska AKS US-Aleutian HAS US-Hawaii HST
# UY-Montevideo UYS VA-Vatican CET YU-Belgrade CET
#
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# NETWORK SETTINGS
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#
dhcp: 1 # 1 = enabled (default), 0 = disabled
# ip: x.x.x.x # set ip address if not using DHCP
# subnet mask: x.x.x.x # set netmask if not using DHCP
# default gateway: x.x.x.x # set default g/w if not using DHCP
# dns1: # set DNS Server IP if not using DHCP
# dns2:
#
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# Network Address Translation (NAT)
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#
# The NAT feature has been added that allows the IP phones to operate while
# connected to a network device that enforces NAT. You can enable or disable NAT
# for a Nortel network, or you can enter a specific NAT IP address and NAT Port
# to use (external IP address and hard-coded port mapped on the NAT device).
#
sip nortel nat support: 0 # 1 = enabled, 0 = disabled
# sip nortel nat timer: 60 # seconds between keep-alives
# sip nat ip: # Assign NAT IP address
# sip nat port: # Assign NAT IP Port for SIP Packets
#
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# Port VLAN Settings
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#
#VLAN is disabled by default. When you enable VLAN, the IP phones uses the default settings for each VLAN parameter
#
tagging enabled: 1 # 0 = false / disabled (default), 1 = true / enabled
#
#——————————————————————————————————————-
# Port 0 Vlan settings
#——————————————————————————————————————-
#

VLAN id: 2 # id assigned to physical port 0: Phone VLAN MUST BE IN CAPS
tos rtp: 32 # 0 to 63 (default=32)
tos rtcp: 32 # 0 to 63 (default=32)
tos sip: 24 # 0 to 63 (default=24)
tos priority map: (24,3)(32,4)(32,4) # DSCP priority mapping for sip, rtp and rtcp (defaults as shown)

#
# This parameter is based on the Type of Service (ToS), Differentiated Services Code Point (DSCP) setting for
# SIP (tos sip parameter), RTP (tos rtp parameter) and RTCP (tos rtcp parameter). It is the mapping between the
# DSCP value and the VLAN priority value for SIP, RTP, and RTCP packets. You enter the tos priority map value as
# follows: (DSCP_1,Priority_1)(DSCP_2,Priority_2)…..(DSCP_64,Priority_64) where the DSCP value range is 0-63
# and the priority range is 0-7. Mappings not enclosed in parentheses and separated with a comma, or with values
# outside the ranges, are ignored.
#
priority non-ip: 5 # priority for non-ip packets. range 0-7 (default = 5)
#
#——————————————————————————————————————-
# Port 1 Vlan settings
#——————————————————————————————————————-
#

VLAN id port 1: 3 # id assigned to physical port 1: Computer VLAN MUST BE IN CAPS

#
QoS eth port 1 priority: 0 # range 0-7 (default = 0)
#
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# Extensible Markup Language (XML)
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#
# The XML application for the IP phones allows users to create custom services
# they can use via the phoneï¾’s keyboard and display. These services include
# things like weather and traffic reports, contact information, company info,
# stock quotes, or custom call scripts.
#
xml application URI: http://10.101.6.250/test.xml # full uri of xml file
xml application post list: 10.101.6.250 # ip address of server/s
xml application title: test.xml # customises the name displayed under the services menu
#
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# Global SIP SETTINGS
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#
sip proxy ip:
sip proxy port:
sip registrar ip:
sip registrar port:
# sip outbound proxy:
# sip outbound proxy port: # Set Port to 0 to perform SRV lookup
sip digit timeout: 4 # default = 4 seconds
sip registration period: 3600 # default = 3600 seconds
sip registration retry timer: 1800 # 30 – 1800 seconds (default = 1800)
#
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# Advanced Global SIP SETTINGS
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#
sip explicit mwi subscription: 1 # 0 = disabled (default), 1 = enabled
#
sip session timer: 0 # setting a value in seconds enables support of RFC4028
# # (0 = disabled)
#
sip T1 timer: 500 # Timer 1 is a SIP Transaction layer timer which given an estimate
# # of the round-trip time (RTT). (Default is 500 msec)
#
sip T2 timer: 4 # Timer 2 represents the amount of time a non-INVITE server
# # transaction takes to respond to a request. (Default is 4 seconds)
#
sip transaction timer: 4000 # time that the phone allows the callserver to respond to SIP
# # messages. Range is 4000 – 64000 msec. (default = 4000 ms)
#
sip transport protocol: 0 # UDP(1), TCP(2) or both(0) for sip messaging. (default = 1)
#
sip rtp port: 3000 # For RTP packets (default = 3000)
#
sip use basic codecs: 0 # 1 = limit codecs to G711 and G729, 0=disabled (default)
#
sip out-of-band dtmf: 0 # 0 = disable, 1 = force RFC2833 (default)
#
# sip customized codec: # specify a prefered codec list using the following options :
# # payload 0 = G.711 ?-Law, 8 = G.711 a-Law, 18 = G.729a
# # ptime (in milliseconds) 5, 10, 15, 20……..90
# # silsupp on, off
# # example: sip customized codec: payload=8;ptime=30;silsupp=off
#
sip dtmf method: 1 # 0 = RTP, 1 = SIP INFO, 2 = BOTH (default = 0)
sip silence suppression: 1 # 0 = disabled, 1 = enabled (default)
#
sip send mac: 0 # Adds an “Aastra-Mac:” header to the SIP REGISTER messages, where
# # the value is the MAC address of the phone.
# # 0 = false (default), 1 = true
#
sip send line: 0 # Adds an “Aastra-Line:” header to the SIP REGISTER messages, where
# # the value is the MAC address of the phone.
# # 0 = false (default), 1 = true
#
sip cancel after blind transfer: 0 # Forces the phone to use the Blind Transfer method available in
# # software prior to release 1.4. This method sends the CANCEL
# # message after the REFER message when blind transferring a call.
# # 0 = false (default), 1 = true
#
sip update callerid: 0 # Enables or disables the updating of the Caller ID information
# # during a call.
# # 0 = false (default), 1 = true
#
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# SIP LINE SETTINGS
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#
# The 480i can support up to 9 lines – 4 via hardkeys and 5 via softkeys
# The 9113i supports up to 3 lines – via hardkeys
# The 9112i support 1 line only
#
# The Hardkey Line keys can be configured globally, using the Global SIP Line
# config, or individually, using the SIP per-line SIP configs.
#
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# Global SIP LINE SETTINGS
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#
sip screen name: Joe Smith # the name display on the phone’s screen
sip user name: 4256 # the phone number
sip display name: Joseph Smith # Caller ID. (display name field of the From SIP header field)
sip vmail: *78 # the number to reach voicemail on
sip auth name: jsmith # account used to authenticate user
sip password: 12345 # password for authentication account
sip mode: 0 # 0 = generic, 1 = BW SCA, 2 = Nortel, 3 = BLA
sip bla number: # Bridged line appearance number
sip proxy ip: proxy.aastra.com # IP address or FQDN of proxy
sip proxy port: 5060 # port used for SIP messages on the
# proxy. Set to 0 to enable SRV lookups
sip registrar ip: aastra.com # IP address or FQDN of registrar
sip registrar port: 0 # for SIP messages on the registrar
sip registration period: 3600 # registration period in seconds
#
ring tone: 0 # 5 tones available using values 0 – 4 (or 5 for silent ringing)
tone set: uk # options = Australia, Europe, France, Germany, Italy, UK or US
#
#
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# SIP Per-Line SETTINGS
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#
#line N # N = line number
sip lineN screen name: 1004
sip lineN user name: 1004
sip lineN display name: 1004
sip lineN vmail: *104
sip lineN auth name: 1004
sip lineN password: 1004
sip lineN mode: 0 # 0 = generic, 1 = BW SCA, 2 = Nortel, 3 = BLA
sip lineN bla number: # bridged line appearance number
sip lineN proxy ip: 10.50.10.102
sip lineN proxy port: 5060
sip lineN registrar ip: 10.50.10.102
sip lineN registrar port: 5060
sip lineN registration period: 60
sip lineN outbound proxy:
sip lineN outbound proxy port: # default = 0
sip lineN dtmf method: 0 # 0 = rtp (default), 1 = SIP Info, 2 = Both
sip lineN ring tone: 0 # 5 tones available using values 0 – 4 (or 5 for silent ringing)
#
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# Configuration Server Settings
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#
# This section defines which server the phone retrieves new
# firmware images and configuration files from. Three protocols are
# supported TFTP, FTP and HTTP
#
download protocol: TFTP # valid values are TFTP (default), FTP and HTTP
# tftp server: 10.60.1.57 # set TFTP server if not provided by DHCP
# alternate tftp server: 10.60.1.56
use alternate tftp: 0 # 1=true 0=false
# alternative tftp server:
#
# ftp server: 192.168.0.131 # can be IP or FQDN
# ftp username: aastra
# ftp password: 480iaastra
#
# http server: bogus.aastra.com # can be IP or FQDN
# http path: firmware # location of firmware.st file
#
directory 1: company_directory # name of directory 1 file on config server – TFTP download only
directory 2: personal_directory # name of directory 2 file on config server – TFTP download only
#
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# Automatic updates and config synchronisation
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#
auto resync mode: 0 # 0 = disabled (default), 1 = configs, 2 = firmware, 3 = both
auto resync time: 00:00 # time for resync in hh:mm
#
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# SIP DIAL PLAN SETTINGS
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#
# As you dial a number on the phone, the phone will initiate a call when :
#
# (1) The entered number is an exact match in the dial plan
# (2) The “#” symbol has been pressed
# (3) A timeout occurs
#
# The dial plan is a regular expression that supports the following
# syntax:
#
# 0,1,2,3,4,5,6,7,8,9,*,# : matches the keypad symbols
# x : matches any digit (0…9)
# + : matches 0 or more repetitions of the
# : previous expression
# [] : matches any number inside the brackets
# : can be used with a “-” to represent a
# : range
# () : expression grouping
# | : either or
#
#
# If the dialled number doesn’t match the dial plan then the call
# is rejected.
#
sip digit timeout: 4 # set the inter-digit timeout in seconds
sip dial plan: “x+#|xx+*” # this is the default dial string, note that this must be quoted
# # since it contains # a * character
#
# Example dial plan……
# sip dial plan: [01]xxx|[2-8]xxxx|91xxxxxxxxxx
# accecpt any 4 digit number beginning with a 0 or 1, any 5 digit number
# beginning with a number between 2 and 8 (inclusive) or a 12 digit number
# beginning with 91
#
# sip dial plan terminator: 0 # 0 = disabled, 1 = enabled
# # disable to use # in the dial string
#
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# softkey & prgkey SETTINGS
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#
# The 480i can support up to 20 softkeys.
# The 9133i supports 7 PrgKeys
# The 9112i supports 2 PrkKeys
#
# Use the following 480i syntax and replace “softkey” with “prgkey” for 9133i and 9112i models.
#
#===================================================================================================================
#
# softkey types: “line”, “speeddial”, “blf”, “list”, “dnd”, “xml”, “flash”, “park”, “pickup”, “empty”
# (note: “blf” and “list” are NOT applicable on the 9112i)
#
# softkey label: Alpha numeric name for the softkey. The maximum number of characters for this value is 11
# for speeddials and 9 characters for lines and blf.
#
# softkey value: Phone number for “speeddials”. Extension to be monitored for “BLF”. No value required for
# “line”, list” and “xml” line types. See 1.4 Admin Guide, Pg. 74, for call park and pickup
# values.
#
# softkey line: This is line associated with the softkey.
#
# softkey states: “idle”, “connected”, “incoming”, “outgoing” – see Pg. 183 of 1.4 Admin Guide for full
# details.
#
# Softkey and prgkey examples…….
#
softkey1 type: speeddial
softkey1 label: “Joe Bloggs”
softkey1 value: 123456
#
softkey2 type: speeddial
softkey2 label: “VMail”
softkey2 value: *500
#
prgkey1 type: speeddial
prgkey1 value: 555
#
# DND Key
softkey3 type: dnd
softkey3 label: DND
#
# Line appearance
softkey4 type: line
softkey4 label: “Sales”
softkey4 line: 5
#
prgkey2 type: line
prgkey2 line: 6
#
# blf
softkey8 type: blf
softkey8 label: Jane Doe
softkey8 value: 4559
softkey8 line: 1
#
# XML
prgkey3 type: xml
prgkey3 value: http://10.101.6.250/test.xml
#
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# Manual audio adjustments
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#
# each of the following parameters can be adjusted by +/- 10db
#
headset tx gain: 0
headset sidetone gain: 0
handset tx gain: 0
handset sidetone gain: 0
handsfree tx gain: 0
#
# Example 1 – handset tx gain: -5 (decrease the transmitted audio gain by 5 db)
# Example 2 – handset sidetone gain: 5 (increase the local sidetone by 5 db)
#
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