TECHNOLOGICAL INSTITUTE OF TIJUANA *** *** Network Operating Systems "Project: PBX" Room 305 Series: 9L1B Professor: M.C.

Rene Solis Date: April 2006 Members: Barragán Corte Tebari Enrique Zamudio Crisanto Muñoz Mario Alberto Fuen tes Luis Miguel Casillas Antonio Álvarez Blanco Cabuto Román Martínez Mata Artur o Arce Fuentes Ulises Romo Mariscal Andres Aldana Ibarra Alvaro Orozco Hernández Joel Sigifredo Documents: A, B Holly Valdez Rios Cintia Torres Lourdes Berenice Gonzalez Alcala Elizabeth C, K Antonio Renteria Girón González Vázquez Miguel A lfonso Rincón Centina F Marco Antonio Valdés Manuela Fernández Item Regalado Kar la Judith G Norma Denisse León González Gloria D Cruz Sánchez Felipe de Jesus Mu noz E Climbing Item Karla Meza Rocio Pelayo Espinoza Astrid Brenda J Arreola R B arrios Tania H, I González Abelardo Solano Lopez Miguel Angel Vargas Monica Patr icia Toledo OLAC BPX Project Introduction Using a PBX PBX phones do not connect all of a company separately to the local p ublic telephone network PSTN, avoiding his time you have to have a separate line with monthly charges and outgoing calls to the telephone central back again to communicate internally. As much as the fax, or modem, or groups of phones, or ot her communication devices can be connected to a PBX (although the modem may degr ade the quality of the line to the modem.) And usually these devices are listed as extensions. The PBX device is installed in the company often requires the ser vice and connects calls between the telephones installed there. It also has a li mited number of lines available for calls outside the site. Companies with multi ple sites can connect their PBXs together via trunk lines. PBX service can be pr ovided from a computer located in dispatching the service provider through the l ocal public telephone network switching. Calls out in a PBX are made by dialing a number (usually 9 or 0) followed by external number, then a trunk line is auto matically selected and it completes the call. Avaya, Alcatel, Comdial, Cisco, Er icsson, Fujitsu, Intertel, Mitel, NEC, Nortel, Panasonic, Samsung, Siemens, Shor eTel, Toshiba, Vertical and Vodavi and others offer PBX equipment. Some features available in a PBX are: 1] Call Transfer 2] system to check the status of exten sions 3] queuing system: Makes someone calls a busy extension, the system to wai t for the caller to that the extension is free (that they put a repetitive jingl e) 4] Conferences, which allows calls from abroad come to speak to several exten sions at once, etc. 1] Keep a file with information on communications 2] system passwords 3] Divert calls at the request of users, if they are to move from his position, etc. A PBX is actually used for members of the organization to which corresponds the PBX t o communicate easily between them, even if they are away from their usual job. T his provides a telephone number that accepts incoming calls, and after requestin g a code (or code without asking), we kindly ask that we press the extent to whi ch we wish to call. From within the PBX, you can also make calls abroad. Long ag o, there were companies that had their PBX fully public, nine are accessed by pr essing the outside line, until they realized that some people who were not calle d your company from the PBX. In a PBX will be at least an outside line, to be po sitioned so that users can communicate with the outside. And at least there will be a line from the outside, so that the outside can communicate with users. Let us focus now on the most modern PBX: Computerized Branch Exchange (CBX) or just Computer Based PBX To better understand the operation of a PBX can do, these ar e some of their bases. The modern PBX combine a computer, a mass data storage, a nd a line switching system. [1] To produce a detailed invoice file to identify a

nd deal with [2] Combine voice communication circuits (day) and fast data commun ications (nights) [3] Manage email communications capacity of a PBX has be caref ully studied, because it has to support communications between extensions. A cri tical factor is the number of logs and links the system. Introduction Every good self-respecting geek has ever thought about converting y our boring desk phone in a smarter unit.€Asterisk is a software (free) very simp le to use (One of their flavors, Asterisk @ Home comes as a live-cd self-configu ring) and is the ideal solution to build a small PBX for your home or office. Wh at we can assert Asterisk? For example, to manage internal extensions and make c alls without going through the telephone operator to give intelligent telephony services using voice recognition for automatic informative phrases ... Who has n ot wanted to hear your favorite song to put a call on hold ? All this and much m ore, with minimal cost, thanks to Asterisk. History · · · · The Asterisk project began in 1999 when Mark Spencer decided to start her own unit because he could not buy one. Seeing the success of Asterisk, founded Linux Support Services at t he end of that year. Linux Support Services becomes Asterisk in 2002. Currently, about 300 developers participating in the development of different modules. Asterisk is an open source implementation of a telephone exchange (PBX). Like an y PBX, you can connect a number of phones to make calls to each other and even c onnect to a VoIP provider or to an ISDN basic and primary. Asterisk is GPL. Mark Spencer of Digium originally created Asterisk and is now its principal develope r in conjunction with other programmers have contributed to correct errors, add new features and functionalities. Originally developed for the Linux operating s ystem, Asterisk now also runs on BSD, MacOSX, Solaris and Microsoft Windows alth ough the native platform (Linux) is the best supported of all. Asterisk includes many features previously only available in expensive proprietary PBX systems: v oice mail, conferences, IVR, distribution automatic call, and many more. Users can create new functionality by writing a d ialplan in Asterisk scripting language or by adding modules written in C or any other programming language supported by Linux. To connect standard analog phones need a FXS or FXO telephone cards manufactured by Digium, or by other manufactu rers, since the server to connect to an outside line is not worth a simple modem . Perhaps most interesting is that Asterisk supports many VoIP protocols such as SIP, H.323, IAX. Asterisk can interoperate with IP phones to act as a registrar and as a gateway between them. Telecommunications companies worldwide are begin ning to use Asterisk as native VoIP system with SER SIP Express Router instead o f other brands that offer proprietary PBX like Alcatel, Cisco and Avaya. Telepho ny solutions based on Asterisk offer not only classic features of a telephone ex change, but also some advanced features that are possible through the integratio n of telephony and computing: · · · · · VoiceMail conference calls over two Call waiting, transfer search in the databas e recording the parameters of the calls The main services integrated into Asterisk include: · · · · · · Password protection by default recording messages and custom messages different messages depending on whether your PC is busy or you do not respond by e-mail No tification of Voice mails get visual and audible message waiting Supplemental Co nference Call an unlimited number of participants is · Vocal server etc.

Interoperability with other systems over IP Voix: Thanks to Inter Asterisk Excha nge protocol (IAX), Asterisk allows interconnection gateways to deploy tradition al telephony as well as to other IP telephony protocols. Towards the other VoIP protocols: · · H.323 Media Gateway Control Protocol (MGCP) Toward the PSTN telephone: · · · · · · · · Robbed Bit Signaling Types FXS and FXO Loopstart groundstart Kewlstart E & M E & M Wink Feature Group D Towards ISDN telephone: · · · · · Lucent 5E DMS100 4ESS EuroISDN the National ISDN2 · BRI (ISDN4Linux) Codecs supported by Asterisk: · · · · · · · · GSM G.729 (available-through purchase of commercial -* Linear Mu-Law G.726 A-Law ILBC LPC-10 MP3 (decode only) The advantages of having a completely software-based PBX are immense, especially if we talk about the quality and robustness of Asterisk: § § § § § A single voice and data cabling. Mobility jobs within your company without havin g to rewire. Setting the unit fully documented software. So cheap you can have a backup. Additions to the second€and without "buy another wardrobe ..." Native Issues Asterisk IP Voice suffers, like the rest of the solutions of the problems of VoIP: - Latency - Jitter - Bandwidth Some can be solved to some e xtent: - Change dynamic supplier base its latency. - IAX2 Trunking to save some bandwidth on the links interasterisk. Configuration problems (complexity) Aste risk is typically configured as text files, the syntax may be slightly 'dull'. There are many managers and user interfaces to configure, but not completely pu rified to be integrated together. Asterisk as a PBX (PBX) / IP PBX Asterisk as a telephony server platform Transparent to Asterisk as VoIP gateway The IP address can be any valid IP address that has Internet access, that is: · · private · May be behind a NAT. It can be approved. May be assigned by a DHCP. It can be assigned manually. Traditional Telephony traditional telephone network is known as PSTN stands for Public Switched Telephone Network or Public Switched Telephone Network and uses circuit switching technology to transmit calls. Establishing a dedicated connect ion or circuit that connects the two parties involved in communication. When you dial a phone number dedicated path is generated from the telephone caller to th e called party. The telephone network provides real-time transmission with guara nteed quality of service ensured by the dedicated circuit during the call. The c

ircuit is not used efficiently because it is dedicated during the entire duratio n of the call, but most conversations are mainly composed of silence, so the cir cuit in use, not broadcasting anything really. IP Telephony IP Telephony SIP as the new standard has nothing to do with conventional forms of Internet telephony . SIP enables phone calls with a high comfort, low cost and with many additional features compared to conventional telephony. Definitions Protocol (SIP Session Initiation, Session Protocol Initation in Engl ish) The initials SIP mean Initiation Protocol Session (Session Initiation Proto col in English), an open, standards-based protocols for telephone calls over IP voice (VoIP) and other multimedia sessions or text, such as instant messaging, v ideo, online gaming and other services. (For example: Sipura SPA 841). SIP phone software: These phones are just one software and can be installed on y our computer or laptop. This solution is free and allows phone calls in a very c omfortable through a headset. Requirement: The computer must be on, if you want to receive calls. If you will not receive them, you can activate the call forwar d with Follow-me. (For example: X-Lite) SIP Adapterbox: The ideal for the home. With the Adapterbox can still use your a nalog phone. Just plug the Adapterbox the broadband line and the analog phone, a nd you can talk on the phone (For example: FRITZ! Box Fon WLAN). VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony call to a group of resources that enable voice over t he Internet travel emplenado your Internet Protocol or IP protocol. In general, this means sending voice in digital form in packets rather than sending in the f orm of circuit switching as a conventional PSTN telephone company. The main adva ntage of these services is that it avoids high telephone charges (mainly long di stance) for ordinary companies. At present the quality of voice is indistinct be tween a VoIP call or a conventional call. Another type of service you can avoid paying a penny in terms of phone calls is to implement a solution within an enterprise VoIP interconnecting branches. Mobi le broadband IP hard Standalone A broadband hard phone is a cell autonomous IP t hat looks like a conventional phone but instead a conventional telephone support , has an Ethernet port through which it communicates directly with a server VoIP , VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone does not require any personal computer or to any software running on a personal computer to make or receive VoIP phone calls. It can be used independently€all that is required is an Internet connection. While solutions based PC software are cheaper, a hard phone is the best solution for IP telephony. Analog Telephone Adapters adapter most common analog phone is a device with at least one telephone port (FXS port) used to connect a conventional telephone and an Ethernet port used to connect t he adapter to the LAN. Using such an ATA it is possible to connect a conventiona l telephone to a remote VoIP server. The ATA communicates with the remote VoIP s erver using a VoIP protocol such as H.323, SIP, MGCP or IAX and encodes and deco des the voice signal using a voice codec such as ulaw, alaw, gsm , ILBC and othe rs. Since ATAs communicate directly with a VoIP server, do not require any softw are to operate on a personal computer, such as a soft phone. IPDID local service number IPDID provides a solution for the card companies for telephone, voice me ssaging / fax service providers, and businesses that require flexible and profit able local phone numbers in several calling areas. The local service IPDID numbe r also provides a solution for bulk phone numbers in a specific area calling. Wi th over 1,000 rate centers in 43 markets to choose from, IPDID service delivers these features: · · National coverage without high contract fees. Flexible Prici ng incoming calls per minute charge outside. Delivery of VoIP without the need f or multiple bursts.

· · Subscriber fixed charges allowing controlled network costs. Portability of numbe rs and help IPDID LNP for existing numbers. Did (Checked Incoming Direct, Direct Inward Dialing in English) Ability to make a phone call to an internal extension without having to go through the operator. Components: Voice Path: The ways of representing a physical port voice / virtua l on a switch and / or data circuit (T1) in which a call is passed. Each voice p ath from one point of view of capacity, are the IP equivalent of a physical line telephone. A path can only support one voice call, multiple paths can support m ultiple voice calls. For example: Your physical T1 to PBX can only support 23 si multaneous calls. If you would like to match that capacity you would need a voic e paths of the order 23 (order 23). As the channels on a T1 determine their phys ical ability to call a voice path determines the IP call capacity. NPA-NXX: The United States is segmented into various Numbering Plan Areas (NPAs), each identi fied by a code of three-digit NPA, commonly called an area code. The NXX identif ies the change assigned central office within the NPAS. When placed together the NPA-NXX pair indicates that the location of the phone number is 678-460 as an N PA-NXX for Atlanta, GA. Each NPA-NXX is sold in a block of 20 issues. Block Number: The phone numbers are purchased in large groups contiguous. Blocks of numbers to call these groups. The minimum size block for NPA-NXX numbers is 20. For example (678) 460 1100 - 1119 represents a block of phone numbers. ATA (Advanced Technology Attachment) The controls ATA mass storage devices data, such as hard drives and ATAPI (Advanced Technology Attachment Packet Interface) devices such as adds, CD-ROM drives. ATA and ATAPI atachment means AT, ATA pack et interface. The various versions of ATA are: • Parallel ATA or ATA. or ATA2. S upports fast transfers and multiword DMA block. or ATA3. ATA2 is reviewed. or AT A4. known as Ultra-DMA or ATA-33 MEDIATYPE wire transfers at 33 MBps. or ATA5 or ATA/66. Originally proposed by Quantum for transfers at 66 MBps. or ATA6 or ATA /100. Support for speeds of 100MBps. or ATA/133. Support for speeds of 133MBps. Serial ATA. ATA remodeling with new connectors (power and data), cables and powe r supply. Acronym for Private Branch eXchange or Private Business eXchange. It i s a telephone that is used for private business. Compared to a telephone company .

The FXS interface provides power (battery) and generates a ring signal. FXS Module for use with the Wildcard TDM400P. The FXS module allows the TDM400P card end analog phones. Because the design is modular, a user can activate addit ional ports at any time with more cards or FXS daughter FXO. The FXS module passes all the call features any standard analog telephone s upport. The FXO interface receives power (battery) and receives a ring signal FXO Module for use with the Wildcard TDM400P. The FXO module allows the TDM400P card complete analog telephone lines (POTS). Due to the modular design, a user c an activate additional ports at any time with daughter cards more FXS or FXO. Th e FXO module passes all the call features any standard analog telephone line wil l support. The global certifications are pending. IP Phone: IP PhoneTM is the right solution for companies where just one or two p eople should do ILD call. The computer connects to a node in the local area netw ork in the same way that any computer.

     

The PhoneTM IP is compatible with the services of internet access high speed, su ch as cable or ADSL. Development To use a PBX service need a broadband Internet (for example Prodigy Infinitum, i -Go, Maxcom, cable, etc.). You can buy a computer with two phone ports to perfor m two simultaneous calls, purchase one or two accounts and one or two phone (s) analog and a node on your LAN Ethernet RJ-45, assign an IP address to your devic e via DHCP or manually and start saving money. The basis for having a PBX, we ne ed a server, in this case is Asterisk. After installation, we get an IP for our server. Then be made about tests: a) b) c) d) e) f) g) h) i) j) k) l) IP - a - I P (call) SIP - a - SIP (G.727 protocol ) IP Fax - to - IP Fax (ATA - ATA) SIP a - SIP (with a vendor Example Sip Phone) Did Virtual number (PSTN - to - SIP) A sterisk - a - SIP (supplier) Asterisk - a - SIPx It will return from the Easter holidays. FXO - Get a FXO PSTN (Asterisk - Asterisk) or (x100p.com) FXO - FXO Ca sar the virtual DID with Asterisk and DID call -> Asterisk Extension Fax - Aster isk Fax SH test

Next Steps: is necessary that the machines are on the same IP addr ess range, in this case we use the following addresses: 192.168.0.2, 192.168.0.3 . When you connect the soft phone, you need to download the drivers, we download ed from windows update. Install SIPPS (and record). In the soft places (SIPPS) r ecords the IP address of the machine to which you want to call. Teams travez con nected to a crossover cable. Configure IP phones (which are necessary). Mark, wh ether soft phone (SIPPS) to hard phone or vice versa. In this case, use the code c that comes by default which is the PCM0. b) SIP to SIP Requirements: Soft Phone (SIPPS) cross Hard Phone Cable Steps: • It is nec essary that the machines are on the same IP address range, in this case we use t he following addresses: 192.168.0.2, 192.168.0.3. When you connect the soft ph one, you need to download the drivers, we downloaded from windows update. • Inst all the SIPPS (and record). • In the soft places (SIPPS) records the IP address of the machine to which you want to call. Login to hard phone settings through Explorer http: \ \ 192.168.0.160 • After clicking on advanced options. Change codecs G729a / B IMAGES: c) IP Fax - to - IP Fax (ATA - ATA) Servers that were installed were connected to a telephone line Objective: To ach ieve communication between two fax, via ip. Hardware Used: - 2 Fax - ATA - UTP c abling. Software used:-ATA Firmware First and foremost, you connected the appropriate cables, configuring the ATA, t his was done entering the ATA menu to configure the software, in that the server ip was 198 168 100 200, fax to each was assigned an address ip and through the ATA was given an extension, one was 11 and the other 12 After a long day of testing different settings by a fax was sent to check whethe r everything worked correctly and it was. These were the Fax that were used for testing. d) SIP - a - SIP (with a vendor Example Sip Phone)

 

a) IP IP Requirements To perform this practice we use the following material: Software: Soft Phone (SIPPS) Hardware: Cable Cross Hard Phone But it also could have used two soft sets.

 

 

 

 

 

 

 

               

We will use a SIPphone Gizmo Project as a supplier and a client (softphone) and is the official supplier, will follow the steps: Step 1: We go to www.sipphone.c om click on member login 2nd step: We will create our own click where it says Create an account 3. shows the confirmation page data 4th step: wait for you in our email accounts e-mail to activate our phone sip ac count. 5th. Step: we hope to send us information that we use. The 6th step: downloaded gizmo http://www.gizmoproject.com/ Page 7th. Step: We choose the platform which you can use and install. We signed and w e scored a number of known and ready. e) Did Virtual number (PSTN - to - SIP) As buy the DID. 1) access to the page siphone.com 2) I create an account of SIP-Phone where you assign a SIP phone 3) Once the account is already access to the area of mobile virtual 4) United States was selected, the area and phone. 5) pay 12 dlls. With a credit card to obtain service only receive calls to your local San Diego 6) G-Proyect under Gizmo, was discharged and password. It was Sa n Diego to test DID. Setting the phone to use a DID. 1) Enter the configuration of the SIP-PHONE. 2) configured to use the provider's server. (SIP Server) 3) Change the user name wh ich is the virtual number of SIP-PHONE Example. 17476485934 (sip-phone service) 4) Set the password. Example. ********* 5) Call the DID number. 0016193780948 f) Asterisk - a - SIP (supplier) Objective: Connect servers from different vendors (SIPphone Asterisk). Software: Software Asterisk VoIP switch. Hardware: Asterisk server. SIPphone Cable Cross Steps: 1. SIP clients are configured to connect to the GATEWAY ACB.getmyip.com (range connections are crea ted at 500). 2. configuration setting in the Outbound SIP Trunks Caller Id, is c onfigured with a SIPphone number. 3. In peer details: o Put server or Password o r Type-per 4. On the record: a. Register String: 17476461850 ß is reported to this number. 5. We tested a provider SIPphone Asterisk extension 500 at: 17,476,461,850 of SI Pphone. 6. It makes the call. IMAGES g) A SIPX ASTERISK AFTER HOLIDAY. h) FXO - PSTN Objective: Make calls from Asterisk to a conventional telephone or a cell phone. This should be used or hardpfhone sofphone. Definitions; One Card FXO (Foreign Exchange Office, in English) is a computer device that connects it to the PSTN, and using special software, make and receive phone calls. Serves mainly to imple ment telephone exchanges (PBX) with a computer. Cards to connect a phone to a co

 

 

 

 

mputer are called FXS. Switched Telephone Network (PSTN Also called Basic Telephone Network or PSTN) te lephone network is designed primarily for voice transmission, but can also carry data, for example in the case of fax or Internet connection through an acoustic modem. The telephone terminals communicate with a communication unit for a sing le channel shared by the microphone and the signal that goes to the headset (the re is only one signal on the wire at one time made up the climb but the descent) , for what are needed echo suppressors. The voice is baseband, ie, without modul ation (the signal produced by the microphone is placed directly into the cable) control signals (call / hang up) is modulated on the same channel with special t ones, which produced that noise calls could be cut. Currently no longer the case with digital exchanges. To access the PSTN from a PC FXO card required, while a nalog phones can communicate with computers with FXS cards .. Necessary requirem ents: Hardware used: hardphones ATA FXO Asterisk Fax Server Microphones Siphon A sterisk Softphone Software used Procedure for conducting this test: Before using SIPPS software from should make some configurations. First you must indicate which type of connection to us e, and indicate the connection speed The next step is to register your details i n order to use the program, the registration screen shown below.

This screen should adjust the volume of microphone and speakers, and can also be testing.

This screen is enhanced to the configurations and finishes.

This is the application that is used to make calls.

This screen is making a call, to make dial 9 first and then the home phone numbe r in case of dialing a cell phone must be from the 044 and the number.

This is the screen Asterisk server, you go to the option of Trunks, and you can see the next screen we leave the option of Trunk ZAP/g0

In outbound caller id you give the line number from which this call reliza you'r e there to outbound routing to verify that the trunk SEQUENCE is the same number as the zap. i) Get an FXO (Asterisk - Asterisk) or (x100p.com) FXO - FXO Objective: Send and receive calls through two Asterisk servers needed Requiremen ts: Hardware used: hardphones ATA FXO Asterisk Fax Server Siphon Microphones Asterisk Softphone Software used Procedure for conducting this test: Before using SIPPS software from should make some configurations. First you must indicate which type of connection to us e, and indicate the connection speed The next step is to register your details i

       

           

n order to use the program, the registration screen shown below.

This screen should adjust the volume of microphone and speakers, and can also be testing.

This screen is enhanced to the configurations and finishes. This practice also should have two Asterisk servers configured so that they are switched. This is the application used to send and receive calls.

This screen is making a call, to make her first dial 9 and the phone number to b e dialed, the line would be being routed. For calls to another phone Asterisk al so has to dial a 9 and the phone number.

This is the screen Asterisk server, you go to the option of Trunks, and you can see the next screen we leave the option of Trunk ZAP/g0

In outbound caller id you give the number you want to flag for you're there to o utbound routing to verify that the trunk SEQUENCE is the same number as the zap. j) the virtual Casar DID with Asterisk and DID call -> Extension Objective: To be able to make calls from a Virtual DID to an extension with aste risk server. What do I need?: To start with you need to find your sip number t o log in How do I find my Gizmo SIP number? You encontarar Sip your number on th e screen where you edit your profile

Then you need to enter your account Asterisk Management Portal, and this start i nstalling Asterisk @ home. There are two sections that needed to be configured o n the Asterisk Management Portal, in order to connect your Asterisk Server with Gizmo Project. Need to create a SIP Trunk is available for your Asterisk box to register with G izmo Project Server. An outbound route needs to be configured, so calls will be sent to Gizmo Project Server. What steps should I follow? 1. Aql will need to click Setup Tab located on the upper right corner of your Asterisk Management Portal. Dar Trunk Click o n the navigation bar select "Add Trunk". 2) You need to configure an Outbound Routes your calls. To do this you have to c lick on "Outbound Route" on the left navigation bar for Asterisk Portal manageme nt firm. k) Fax Asterisk - Asterisk Fax Objective: To ensure communication with fax, through a server that has installed Asterisk. Hardware used: - ATA - 2 fax - Wired - A server Software used: - Aste risk - Firmware

 

 

   

         

This procedure was almost equal to ip-ip fax fax with the difference here is use a server that was installed asterisk Conclusions PBX is a telephony system that interconnects telephone extensions to the core domestic telephone (PSTN). The PBX includes advanced routing of calls, call forw arding, conferencing, call waiting, etc. Modern PBXs use digital switching metho ds that can support the installation of telephones and analog and digital lines. The main advantage of the service is the significant savings on international c alls. Apply the same rate at any time of day or night. He has extensive plans an d devices to meet the needs of all users, from home user up to several hundred c orporate extensions .. It all depends on your internet connection and network tr affic at the time you make the call, but we can come on that 90% of the calls yo u make will be of excellent quality almost indistinguishable from the convention al service, as each call only consumes 17kbps bandwidth if you have a broadband connection and several connected PCs that come through this connection you must bear in mind that the total bandwidth is divided among all connected devices and if an ADSL connection is asynchronous means that if "low" to 512kbps speed of " up" will be 256kbps and this is split between all your devices.€Since it is an i ndependent technology works no matter PCs operating system used today. The only thing that would use a PC is to see your balance and / or register and this can be done under any platform and that is through Internet. The PC-Dialer software currently only works under Windows any version.