Sunday, January 31, 2016

Direct Inward Dial trunk

Direct Inward Dial  trunk:  Trunks that  allow  direct  inward dialing  from  the public  network to the Norstar  system. Direct inward system access (DISA):  The  feature  that  allows  remote users to  dial  directly  into  the  Norstar system and use Norstar features. Callers will hear  stuttered dial tone  and will  be required to enter  a  Class of  Service password  to  gain access  to the  system. See Remote  Access. directed  pickup:  See Call  Pickup Directed. Directory number  (DN):  A unique number that is automatically assigned to each telephone or data terminal.  The DN, also  referred to as  an  internal number, is  often used to identify  a telephone when settings are  assigned during programming. Default  DN assignments start  at  221. DISA DN:  The received  number assigned to the  Norstar  direct  inward system access  facility. If a caller dials  a number that  is assigned to the DISA DN, the  caller hears stuttered dial tone  and  must enter a Class of Service password. Once  the password is accepted,  the  caller  hears system  dial tone and can  use  Remote  Access features.  See  Remote  Access. Disconnect Supervision:  A setting that  enables the  Norstar system  to  detect if an external  caller hangs  up.  Once  an external caller hangs up, the  Norstar system can disconnect its line. Disconnect  Supervision  is  enabled under  Trunk/Line  data  in  Lines programming. display:  A liquid crystal display (LCD) on  the telephone that guides you through  feature operation and programming. display button:  Norstar system  twoline  display telephones are  each equipped with  three buttons  located directly beneath  the display.  During feature operation or  programming, some or  all  of these buttons  may be used to provide  further  options. If  an option  is available,  it  is  shown in  the bottom row of the  two-row  display, directly above the  corresponding display button. Display buttons  are represented in this manual as  underlined  capital, such as OK . Distinctive ring:  Lines, telephones, and  hunt group DNs can  be  configured with  a  distinctive  ring that  also determines  the  priority  of a  call. DN:  See Directory  number. Do  Not Disturb:  ≤°fi A feature that  stops calls from  ringing at your telephone. Only Priority Calls will ring at your telephone.  A line button will flash when  you receive a  call, but the call  will  not ring. DTMF:    See  Dual tone  multifrequency. dual  tone multifrequency:   Two distinct telephone  signaling tones  used for dialing. DTI:  See Digital  Trunk Interface. E E&M/DISA Trunk  Cartridge:  The Trunk  Cartridge that allows  you to connect  E&M trunks to the  Norstar system.  The E&M Trunk Cartridge  also allows  DISA  access  to  the  system  by providing DTMF  receivers for autoanswer trunks. Emergency 911 dialing:  The capability to access a public emergency response system by  dialing the  digits 9-1-1. State and local  requirements for support of Emergency 911 Dialing

Thursday, January 28, 2016

Separate ring assignments

Each station in the system can be programmed to provide an audible signal when the system detects an incoming call on specified CO/IP lines.  Separate ring assignments are made for Day, Night and Timed Ring operation mode.  In addition, the audible signal at  the station can be delayed by 1 to 9 ring cycles allowing other stations to answer the call first. Operation System Operation of this feature  is automatic. Conditions 1.  Separate assignments are made for stations to  ring in the Day,  Night  and/or Timed Ring mode. 2.  Audible alerting for an incoming VoIP call is based only on the derived IP Address. 3.  A busy station receives  muted ring  or Call Waiting tones as appropriate  for the  station’s Offhook ring assignment. 4.  The system Ring mode  can be selected manually  or automatically.  In the Automatic mode, Day/Night selection is determined based on the  Automatic Ring Mode  Selection table.  The Attendant has manual control over the Ring mode selection.  5.  The LCD of the Attendant station will display Night and Timed Ring  Mode and the  [DND] button LED will flash. 6.  If a CO/IP line is not assigned to ring at any station, incoming calls on the CO/IP line  will ring the first available Attendant.

Tuesday, January 26, 2016

Ability to use Digital (DKT) and Analog (SLT) phones

Full  Featured:   Full  PBX  feature  set  including  auto attendant,  voicemail,  call transfer,  call  forward,  call  park,  call  pickup,  speed  dial,  station  group,  3-party voice conference  and more • Flexible:   Ability  to  use Digital  (DKT)  and  Analog (SLT)  phones  along  with the  full  selection  of  iPECS  IP  phones and  DECT  phones.  A  wide  range  of mobile applications  and soft  clients are also  supported. • Secure:    Supports  IPSec  and sRTP  security  protocols • Affordable:   Low  TCO  with  simple  and  straightforward  licensing structure • Leading  Technology:    Advanced  IP  networking  with  local  and remote management  through  an  intuitive  HTML5-based GUI

Sunday, January 24, 2016

Total Access 900e

Features and Specifications The  Total Access 900e Series products have the following features: • Support for 4  DS1  (or 3  DS1  plus  1 PRI/CAS,  or 2  DS1  plus  2 PRI/CAS)  interfaces • Support for a single built-in FXO interface • Support for up  to 24  FXS  ports  with  octal FXS daughter board • Support for up  to  16  FXS ports  and  8  FXO ports  with  octal FXO  daughter  board  (Total Access 924e  only) • Supports Primary Rate ISDN (PRI) or Robbed  Bit Signaling  (RBS) on the PRI/CAS interfaces • Support for a two  auto  MDI/MDX 10/100BaseT Ethernet ports (RJ-48C) • Full-featured AOS IP router/firewall • QoS/NAT/DHCP client, server, and relay • Support for SIP trunks • Support for up to 6 Mbps of  multi-link Frame Relay,  multi-link PPP • Support for optional VPN - 500 IPSec tunnels using  DES/3DES/AES encryption • Support for 3-way conferencing • Support for caller ID, message waiting,  and stutter dial  tone • Fax and  analog  modem compatible  (V.90) • Support for local station to station calls • Up to  48  channels  of G.711  (µ-law) • Up to  48  channels  on  G.726 (32K  ADPCM) • Up to 48  channels on  G.729 • Up to  48  channels  of  DTMF detection/generation

Saturday, January 23, 2016

Make a call

MAKE  CALL Description There are three types  of  call  setup  –  Station  Call,  CO  Call,  System  Call  Feature  Implementation. Operation Making Station Call  Setup 1.  Dial  Station Number. 2.  If  ‘Dial Digit  Map’  is  programmed,  SIP  Phone will  send call  setup immediately. 3.  If  ‘Dial  Digit  Map’  is  not  programmed,  press  the  [SEND]  button or  “#”  key  for  send out  call  setup. Making CO  Call  Setup 1.  CO  Access  Code +  Dial  Number  +  [SEND] ex)  CO  Access  Code “9”,  Dial  Number  ‘450-4500’, dial  ‘94504500’  and press  [SEND]  or  “#”  button NOTE If  you program  ‘Second  Dial  Tone Digit  Map’  on  SIP  Phone,  you will  hear  self  dial  tone from  SIP  Phone. CO Access  Code +  [SEND], after  hearing  CO  dial  tone,  press  Dial  Number ex)  CO  Access  Code  “9”,  Dial  Number  ‘450-4500’, dial  “9”  and press  [SEND]  or “#” hear  CO  dial  tone from  system dial  ‘4504500’ Making System  Call  Feature Setup 1.  System  Call  Feature by  Numbering  Code :  System  Numbering  Plan (PGM106-109) 2.  Enblock  Dialing  :  System  Call  Feature numbering  code +  data +  [SEND]  button. 3.  Supported Call  Features  by  Numbering  are, Internal  Page Zones Internal  All Call Page Meet  Me Page Internal  All Call Page Meet  Me Page Internal  All Call Page External  Page Zone External All Call Page All Call Page SMDR Account  Code Enter SLT  Last  Number  Redial Do-Not-Disturb(DND) Call  Forward Speed Dial  Program SLT  Speed Dial  Access

Wednesday, January 20, 2016

CALL Description

MAKE  CALL Description There are three types  of  call  setup  –  Station  Call,  CO  Call,  System  Call  Feature  Implementation. Operation Making Station Call  Setup 1.  Dial  Station Number. 2.  If  ‘Dial Digit  Map’  is  programmed,  SIP  Phone will  send call  setup immediately. 3.  If  ‘Dial  Digit  Map’  is  not  programmed,  press  the  [SEND]  button or  “#”  key  for  send out  call  setup. Making CO  Call  Setup 1.  CO  Access  Code +  Dial  Number  +  [SEND] ex)  CO  Access  Code “9”,  Dial  Number  ‘450-4500’, dial  ‘94504500’  and press  [SEND]  or  “#”  button NOTE If  you program  ‘Second  Dial  Tone Digit  Map’  on  SIP  Phone,  you will  hear  self  dial  tone from  SIP  Phone. CO Access  Code +  [SEND], after  hearing  CO  dial  tone,  press  Dial  Number ex)  CO  Access  Code  “9”,  Dial  Number  ‘450-4500’, dial  “9”  and press  [SEND]  or “#” hear  CO  dial  tone from  system dial  ‘4504500’ Making System  Call  Feature Setup 1.  System  Call  Feature by  Numbering  Code :  System  Numbering  Plan (PGM106-109) 2.  Enblock  Dialing  :  System  Call  Feature numbering  code +  data +  [SEND]  button. 3.  Supported Call  Features  by  Numbering  are, Internal  Page Zones Internal  All Call Page Meet  Me Page Internal  All Call Page Meet  Me Page Internal  All Call Page External  Page Zone External All Call Page All Call Page SMDR Account  Code Enter SLT  Last  Number  Redial Do-Not-Disturb(DND) Call  Forward Speed Dial  Program SLT  Speed Dial  Access

Tuesday, January 19, 2016

VM pilot

TO FORWARD  ALL   INCOMING  CALLS  TO  YOUR  MAILBOX ❍ Press the Speaker  key  ●  Dial  741 or  press the  Call  Forward  Immediate Function  Key  (if one  is  programmed on  the phone)●  Dial  1  to  Set  ●  Dial  the VM Pilot Number  ●  Hang  up TO FORWARD   INCOMING CALLS  TO  YOUR  MAILBOX  WHEN  YOUR  PHONE  IS BUSY ❍ Press the  Speaker  key  ●  Dial  742 or press the Call  Forward  Busy  Function Key (if one is  programmed on  the  phone)●  Dial  1  to  Set  ●  Dial the  VM  Pilot Number  ●  Hang  up TO FORWARD  INCOMING  CALLS  TO  YOUR  MAILBOX  WHEN  YOU  DO  NOT  ANSWER ❍ Press the Speaker  key  ●  Dial  743 or  press the  Call  Forward  No  Answer Function  Key  (if one  is  programmed on  the phone)●  Dial  1  to  Set  ●  Dial  the VM Pilot Number  ●  Hang  up TO FORWARD   INCOMING CALLS  TO  YOUR  MAILBOX  WHEN  YOUR  PHONE  IS BUSY  OR  YOU  DO  NOT  ANSWER ❍ Press the Speaker  key  ●  Dial  744 or  press the  Call  Forward  Busy/No Answer  Function  Key (if  one  is  programmed  on the phone)●  Dial  1  to  Set  ● Dial the  VM  Pilot Number  ●  Hang  up

Monday, January 18, 2016

Initially Logging in to System

Initially Logging in to System Administration............................................... 22Selecting the System Administration Prompt Language............................. 23Programming System Parameters.............................................................. 24Programming the System Language Mode and System Language......... 24Setting the System Language Mode ......................................................25Setting the System Language ................................................................25Programming the Call Answer Service Operator Extension .....................28Programming the General Mailbox Owners............................................. 29Programming the Maximum Extension Length

Sunday, January 17, 2016

Voicemail application

Auto  Attendant  / Voice  Mail  Application –AA  delivers  recorded  announcement  to direct  callers  to  the  proper  destination   –Voice  Mail  includes  message  broadcast, email  and  mobile  notification   –Offers  all  the  common  VM  functionality –Both  provide multi-language  support › Built-in  Automatic  Call  Distribution (ACD) –Flexible  incoming  call  routing –Real-time  agent  monitoring  and  call  record statistics   –Event  messages  for  management  reporting › Mobile  Extension –Allows  the  mobile  to place  and  receive  calls through the  system –Calls  sent  to  a  user’s  iPECS  phone  and mobile simultaneously › Centralized  Control  T-NET (Transparent Network) Central  UCP  controls  all  modules  and terminals  located  in  remote  offices  providing all  the features  and  functions  of  the  central UCP Local  survivability  is  provided  with  a  second call  server  located  at  a  remote  site Power  redundancy  available  when UCP100/600/2400 installed in main cabinet

Saturday, January 16, 2016

iPECS UCP

iPECS UCP is  designed to deliver  the flexibility you  need  as  your organization  grows Simple  Unified Communications  Built-in Integrated  Applications Tailored  to  your  Needs › Users  can  access voice, video,  instant messaging,  conference calling  and  visual voicemail,  all  on  a simple and  easy-to-use platform Leverage the Latest Standards-Based Technologies › iPECS  helps  you  make the  most  of  the  latest network  technologies such  as SIP, optimize call  costs  using  Wi-Fi  or use  built-in  voice conferencing –Provides  capacity  for up to 2,200  devices, allowing  it  to  handle most  any  need iPECS-UCP Anywhere,  Anytime Connectivity › Access  the  power  of your  iPECS  call  servers your  way  regardless  of your  device  or  location using  smartphone,  tablet or PC  applications › iPECS  offers  a  range  of enhanced  applications from  Ericsson-LG  and other  specialist application  providers, including  Microsoft Outlook  or  Lync  as  well as  others Reliable  and  Resilient › Total reliability  is  the only  option  for  your communications.  With inherent  modular architecture,  iPECS UCP  provides geographic  redundancy, hot  standby  power redundancy  and Central Control  T-Net

Friday, January 15, 2016

Installation

Installation/Connection Cabling Category 3 or 5 twisted pair cable is recommended for all Valcom  distributed amplified paging installations.  Screw terminals are provided  for the basic  connections.  RJ45 jacks are provided for chaining multiple  V-9964 units together.  Removing the narrow right side panel of the  V-9964 provides access to  controls, connections and option switches.  To remove the panel, loosen the two screws holding  the panel in place and lift the panel. Mounting The V-9964  may be wall  mounted or  rack mounted in a  standard 19  inch equipment rack using the brackets included. Connections See Figure 1 for a connection diagram. Tip 1, Ring 1 INPUT 1 is the normal Primary or  Call Stacker system input, and connects to a Loop Start Trunk Port, 600 Ohm Page  Port or some  Valcom Page Controls.   Note: Do not connect to a C. O. Line. Control Input 1 Provides  contact closure activation when  using a Page Port. Tip 2, Ring 2 INPUT 2 is the Override page or Call Stacker line two input.  If desired, connect this to  a second  Loop Start Trunk Port or Page Port. Note: Do not connect  to  a C. O. Line. Provides  contact closure activation when  using a Page Port. Background Music Input Connection for external line level music source (Example: V-2952, FM Tuner). NOTE:  If   using multiple V-9964 units in a chained configuration, all speakers must connect to  the output of  the last  unit   in the chain. Line Out Output connections  to Valcom amplified speakers or 70 Volt amplifier Aux input. Loop Out Connects to Tip and Ring  input on a Valcom multi-zone page control unit. Expansion In RJ45 connection from the previous  V-9964 in a chained configuration.   Expansion Out RJ45 connection to the next V-9964 in a  chained configuration.  Closing a switch  connected to pin 7 and pin 3 of  either Expansion In or Expansion Out will hold  all queued recorded messages, on all linked V-9964s.  During  this time, additional messages may still be recorded into the            V-9964(s).  Once the  switch is opened,  all queued messages, including DTMF, will  play in their entirety. Abort To abort a message during play, connect an external relay  contact across the two abort terminals. NOTE: To abort a message during the record sequence, press any  DTMF button on  the dial pad of the access telephone. Relay  Closure Outputs (PLAY)  PLYSW and PLYMK Normally open relay contact that closes  while a message is being broadcast. (RECORD)  RECSW  and  RECMK Normally open relay contact that closes  while a message is being recorded

Thursday, January 14, 2016

Do Not Disturb Function

Do Not Disturb Function The N.E.C key sets have a function so that anyone trying to ring the key set receives a BUSY tone  or routes to voicemail  even if it is not being used. This function is useful if a room is needed for an important meeting and the keyset needs to  be  available  for outgoing calls  but needs to be silent so as not to disrupt the meeting. Pressing the  DND  key then choosing one of the following options does this: 1  -   External calls only 2  -   Intercom calls only 3  –  All calls 0 - Cancel

Tuesday, January 12, 2016

Dial pulse

The system is compatible with 500 type (dial pulse) and 2500 type (DTMF) analog telephone devices.  This includes on-premises single line telephones (SLTs), fax machines, and modems. In DSX-40, SLTs connect to analog ports in the main equipment cabinet. In DSX-80/160, SLTs connect to SLIU PCBs. Each analog port provides power and ring voltage for the connected SLT.  The analog ports use DTMF receivers. Each system provides 10 DTMF receivers that are shared by all connected analog ports. Message Waiting Both DSX-40 and DSX-80/160 support FSK Message  Waiting lamps. DSX-80/160 also provides support for high voltage Message  Waiting lamps  – while DSX-40 does not. Ringing For Incoming Calls Single line extensions ring according to the settings in  2132-[01-64]: Line Ringing Stations: Config: Ring Line Ringing (2132): Ring Assignment] Assign: . It is not necessary to assign single line sets to Ring Groups to make them ring for incoming calls; they follow Key Ring instead. • In DSX-80/160 by default, the  first 16 extensions (300-315) ring (option 2) for lines 1-12 and  flash (option 1) for lines 13-64.  All other extensions have lamp only (no ringing) for all lines. • In DSX-40 by default, all extensions (including single line sets) have immediate ring for all lines. Ringer Equivalence Number (REN) Considerations DSX-40 Single line telephones assigned to Key Ring or the same Ring Group will ring simultaneously.  This is also true for single line telephones connected to the same port. Since the Ringer Equivalence Numbers of connected single line telephones are cumulative, you must do the following: • Add up the RENs of all connected single line telephones. • Be sure the total REN does not exceed 4 on any single port  or  system-wide. Note that a REN of 1 is normal for an industry standard 2500 set with electromechanical ringer. Many phones with electronic ringers have significantly lower RENs. Check the label on the bottom of each single line telephone for the REN value. DSX-80/160 Single line telephones assigned to Key Ring or the same Ring Group ring in pairs according to their SLIU PCB port assignment. For example, ports 1 and 2 ring together, followed by 3 and 4, 5 and 6, and  finally 7 and 8. If the system has more than one SLIU PCB installed, the respective port pairs ring simultaneously on each card (e.g., ports 1 and 2 ring simultaneously on each PCB).  The SLIU provides the capability to support this rin

Monday, January 11, 2016

maximum of 84 trunks are available

Description The system provides flexible routing of incoming CO (trunks) calls to meet the exact site requirements. This lets trunk calls ring and be answered at any combination of system extensions. A maximum of  84 trunks are available. For additional information on making trunks ring, refer to  Ring Groups on page 1-763. Delayed Ringing Extensions in a Ring Group can have delayed ringing for trunks. If the trunk is not answered at its original destination, it rings the DIL No Answer Ring Group (this ring group applies to DIL or non-DIL trunks). This could help a secretary that covers calls for their boss. If the boss does not answer the call, it rings the secretary’s telephone after a programmable interval. Universal Answer Universal Answer allows an employee to answer a call by going to any Multiline Terminal and dialing a unique Universal Answer code. The employee does not have to know the trunk number or dial any other codes to pick up the ringing trunk. You normally set up Universal Answer along with Universal Night Answer (refer to  Night Service on page  1-663). When a Universal Night Answer call rings the External Paging, an employee can answer the call from the first available telephone. You might also want to use Universal Answer in a noisy warehouse or machine shop where the volume of normal telephone ringing is not adequate. After hearing the ringing over the Paging, an employee can then easily pick up the call from a shop telephone. The Automatic Off-Hook Answer of Universal Answer Call options (PRG 20-10-07) determines whether or not the extension has the Auto Answer feature for ringing calls. This option allows a user to simply lift the handset to answer a ringing call; dialing the service code is not necessary. Additional Trunk Ring Tones Various ring tone patterns and melodies for incoming calls are available (PRG 22-03-01);  Ring Tone Patterns 1~8 and Melodies 1~5 (V3.0 or higher)  can be configurable, refer below chart.

Sunday, January 10, 2016

Terminal information

Line Information: Shows the terminal that is going to be used.  You can set the following items: ❒ ❒ ❒ Extension Number: Shows the extension number you installed. If you change the terminal or number please reset the number and reboot the PC in order for the change to take affect. Extension Name: Shows the name of the Extension. Line Configuration: Shows the Line Configuration screen.  You can't press the button unless you are connected to PBX. ❍ Network Information: ❒ ❒ ❒ PBX IP Address: Shows the IP  Address which you entered during installation. If you change the PBX IP  Address within the UX5000 programming, change this setting as well and reboot your PC. PBX TCP Port: Shows the  TCP  Port  Address you entered during installation. If you change the PBX  TCP  Port  Address within the UX5000 programming, change this setting as well and reboot your PC. My Host Name / IP Address: Shows the My Host Name/IP  Address you entered during installation. We  recommend using a "Localhost" (Default) or the PC name. Otherwise, enter the IP address of the network card you are using. If you change the IP  address of the card you’re using, change this setting as well and reboot your PC. ❍ Operation Mode: Shows the Operation Mode you selected during the installation. ❒ Multi Line: Send event to CTI that occurs on Speaker Key/Answer Key/Trunk Key/Virtual Extension Key/Yellow Key. Use this setting if the application shows multiple call information.

Saturday, January 9, 2016

LIne Information

Line Information: Shows the terminal that is going to be used.  You can set the following items: ❒ ❒ ❒ Extension Number: Shows the extension number you installed. If you change the terminal or number please reset the number and reboot the PC in order for the change to take affect. Extension Name: Shows the name of the Extension. Line Configuration: Shows the Line Configuration screen.  You can't press the button unless you are connected to PBX. ❍ Network Information: ❒ ❒ ❒ PBX IP Address: Shows the IP  Address which you entered during installation. If you change the PBX IP  Address within the UX5000 programming, change this setting as well and reboot your PC. PBX TCP Port: Shows the  TCP  Port  Address you entered during installation. If you change the PBX  TCP  Port  Address within the UX5000 programming, change this setting as well and reboot your PC. My Host Name / IP Address: Shows the My Host Name/IP  Address you entered during installation. We  recommend using a "Localhost" (Default) or the PC name. Otherwise, enter the IP address of the network card you are using. If you change the IP  address of the card you’re using, change this setting as well and reboot your PC. ❍ Operation Mode: Shows the Operation Mode you selected during the installation. ❒ Multi Line: Send event to CTI that occurs on Speaker Key/Answer Key/Trunk Key/Virtual Extension Key/Yellow Key. Use this setting if the application shows multiple call information.

Friday, January 8, 2016

Telecommunication Registration

North American  Regulatory Information 21 Safety 21 Enhanced 911 Configuration  22 Telecommunication Registration 24 Network Connection 25 Canada  and US 25 Hearing Aid  Compatibility 25 Electromagnetic  Compatibility 25 Telephone Company  Registration 26 Use  of a Music Source 26 Rights of the  Telecommunications Company 26 Repairs 27 Canadian Regulations - please read carefully 28 Notice 28 Notice 29 US Regulations - please read carefully 30 Federal Communications Commission (FCC) Notice 30 Ringer Equivalence Number 31 Hearing Aids 32 Programming Emergency Numbers 32 EMI/EMC (FCC Part 15)  32 Important Safety Instructions 34 Installation 34 Use 35 International Regulatory Information 37 Safety 39 Additional Safety Information 40 Limited Warranty 42 Exclusions 42 Warranty Repair  Services 43 After Warranty Service 43

Thursday, January 7, 2016

Message wait

SLT  MESSAGE  WAIT  INDICATION Description All  SLT  devices  will  receive a “Stutter”  dial  tone as  an audible  Message  Wait  Indication.  In addition,  industry  standard Message  Waiting  telephones  may  be connected  to the system. Software will  cause the lamp to  flash when a messaging  is  waiting. Operation System The system  switches  the  90 VDC  lamp  On and  Off  for  assigned SLTs  indicating a Message Wait. Conditions 1.  The system  switches  a 90 VDC  supply  On and Off  to  flash the SLT’s  neon lamp. 2.  Although the SLT  Battery  Feed is  removed during  the 90 VDC  On cycle,  the  system will  recognize an SLT  Off-hook  event. 3.  The SLT must  incorporate a 90 VDC  neon lamp that  is  connected directly  across the  tip and ring  of  the voice network.

Sunday, January 3, 2016

Calling phone

The calling phone sends out an invite

The called phone sends an information response 100 � Trying � back.

When the called phone starts ringing a response 180 � Ringing � is sent back

When the caller picks up the phone, the called phone sends a response 200 � OK

The calling phone responds with ACK � acknowledgement

Now the actual conversation is transmitted as data via RTP

When the person calling hangs up, a BYE request is sent to the calling phone

The calling phone responds with a 200 � OK.

It�s as simple as that! The SIP protocol is easy to understand and logical.

Saturday, January 2, 2016

DETERMINING WHICH ADVANCED FEATURES

DETERMINING  WHICH  ADVANCED  FEATURES  ARE NEEDED Now that you  have your basic  ACD Groups  up and  running, you may want to  fine tune their operation. The  chart  below lists the Advanced ACD Features that will help you get the most  out of  your  system. Review each capability  below  and then turn to  the indicated  page if  the feature is something you  need.   Advanced  Features  Guide ACD Capability When  all agents are busy, incoming  calls can route to  other extensions,  ACD  groups  or  Voice  Mail. Set various options  for  ACD Groups. Temporarily log out an  ACD  Agent. See the  status  of  your  ACD Group's  calls at  a glance. Set  up SIE keys  for ACD Groups. Get  one-button  ACD Group  calling and Transfer  as well  as  a  unique  BLF  for  ACD agents. Temporarily busy out your  phone  to  the ACD  Group when  you need to work at your desk. Use a headset for privacy  and convenience, and optionally answer  calls automatically. Analyze system usage and calling patterns.  Overflow  Options  ACD Setup  Options  Off-Duty  Mode Queue Status Display SIE Key  for ACD Groups  Hotline for  ACD  Agents  Wrap-Up  Time  Headset Operation (with Automatic  Answer)  Traffic  Reports