Introduction to VoIP

Transcription

1 Introduction to VoIP Voice over IP or VoIP, consists in transmitting voice over the IP protocol. Said like this it can sound simple, but IP networks were mainly designed for data and many of the advantages of IP networks for data are a disadvantage when it comes to voice. This is because the voice is very sensitive to delays and transmission problems, even if they are small. As such, transmitting voice over the IP protocol required many technical problems to be solved. Fortunately, technology has evolved and due to the abilities of some talented engineers, it has resulted in solutions that remove the need for us to deal with those problems inherent to IP networks that can damage voice quality. The evolution in the quality of voice communications over IP in the last years has been really surprising. Today we can say that voice transmission on the Internet is a viable alternative within the reach of most of us. 1.1 VoIP: a protocol soup There are many protocols involved in voice transmission over IP. There already are network protocols involved such as IP itself and other transport protocols such as TCP or UDP. On top of that, you have voice signaling protocols. If this were not enough, many more signaling protocol options are available, which can make everything sound a bit confusing at first. VoIP protocol soup

2 Classifying VoIP protocols To simplify things we can classify the protocols used in VoIP in three groups: signaling protocols, voice transport protocols and IP platform protocols. Signaling protocols VoIP signaling protocols perform functions that are similar to their counterparts in traditional telephony, such as session establishment tasks, call progress control, amongst others. These protocols are found in the session layer (layer 5) of the OSI model. There are some signaling protocols that have been developed by different manufacturers and organizations such as ITU or IETF, and which are supported by Asterisk. Some of them are: SIP IAX H.323 MGCP SCCP Among the most popular in regards to Asterisk are SIP and IAX. Further ahead in this book, we will go into more details about the SIP protocol. Voice transport protocols We should not confuse voice transport protocols with low-level transport protocols such as TCP and UDP. Here we are referring to the protocol that transports the voice as such or what is commonly known as the packet. This protocol is called RTP (Real-time Transport Protocol) and its function is simple: to transport the voice with as little delay as possible. This protocol starts to function once the signaling protocol has established the call between the participants. IP platform protocols In this category, we will group protocols from basic to IP networks, which form the basis of the addition of other voice protocols. Among these protocols, we can mention Ethernet, IP, TCP and UDP. Note: In the present chapter, we will emphasize the explanation of this group of protocols so we can bring up to speed those without solid knowledge in Networking. However, the other protocol groups will be examined in more detail in a chapter that is specially dedicated to this purpose.

3 Protocols involved in a SIP call. The case of IAX is very similar. In the above figure, we can observe a curious fact in that although SIP supports both UDP and TCP we can only see it on top of UDP. This is not a mistake but it is because until Asterisk 1.4 the SIP implementation is only available for UDP. Starting in Asterisk 1.6 (Elastix 2.0) SIP is available on TCP as well as on UDP. 1.2 IP Protocol IP protocol is without a doubt one of the most popular ever implemented, mainly because of the great network of networks called the Internet that uses this protocol for its routing. For this reason, before we go into detail about IP protocol we will use a pair of paragraphs to summarize the origins of the Internet. The Internet The internet had its origins in the 1960's as a project created by the Department of Defense of the United States with the goal of creating a robust network that could connect some public sector facilities with universities. The idea behind this network, originally called ARPANet, was that if any of the nodes stopped working this would not significantly impair the operation of the network. For this a decentralized network was created that could handle more than one route between two points. In short, it was about building a fail-proof network (let us remember this was during the Cold War) that was also low-cost. Thanks to this goal, ARPANet evolved into what we know today as the Internet, such a big and powerful communications network that has made an impact in our cultures without a doubt.

4 Without its existence, the Elastix project would never have existed and you would not be reading this now. What is IP protocol? IP protocol (Internet Protocol) works at a network level where information is sent in packets called IP packets. This protocol offers a service without guarantee, also known as a best effort service. That is, nothing guarantees that the packets will reach their destination, although the best effort will be made to have them arrive. IP Address An IP address is a unique number, which also gives information on the location of the machine that has it. To ensure the uniqueness of this number, an authority controls the assigning of IP addresses called IANA (Internet Assigned Numbers Authority.) This unique number is binary and has 32 bits. This number is shown as four decimal number separated by periods for practical considerations and for easy understanding by humans. Each of the four decimal numbers can have a value from 0 to 255. An example of an IP address is Translation from decimal notation to binary (32 bits) of an IP address IP packet As we said before, IP protocol is the protocol that divides information into packets, which are sent to its destination. The advantage of having information in packets is that each of these packets can take a different route to reach its destination. That is, there is a redundancy in the number of routes and it is less likely for packets to be lost. Some readers will ask themselves: What do these so-called packets look like? The format of an IP packet is designed to carry information that allows it to be dissected to its destination and obviously to be reassembled with other packets at the destination to recover useful information. The figure below represents an IP packet header.

5 IP routing IP packet header Routing allows the optimal route to be determined so that an IP packet reaches its destination. To know where a single packet wants to go, we have to examine the IP address of the destination in the Destination Address field in the heading of an IP packet. The devices in charge of routing the packets to their destinations are known as routers. They contain routing tables with information on how to reach other networks. That is, when a packet reaches a router, the router examines the IP address of the destination and tries to determine what network this IP address belongs to. Determining the network where a single packet wants to go is not hard because that information is part of the IP address. That is because each IP address can be divided into two parts: the part that identifies the network and another one that identifies the device (also known as the host.) Identifying each of these parts is not a trivial matter since there exist networks of different sizes and the number of bits that correspond to the network and the host vary respectively. The solution to this is an additional parameter known as the network mask. Network mask and network address calculation Let us see then how to obtain the information of to what network does a device belongs to based on its IP address so that a router can correctly send a packet to that destination. The network mask is a 32-bit number like the IP address. It is called a mask because if it is superimposed over the IP address it allows us to identify which part corresponds to the network address and which one corresponds to the host address.

6 Let us imagine that we have the IP address with a mask of To show how the mask is applied to the IP address we will turn this data into binary as is shown in the diagram below. Mask application to obtain the network and host information Now we can see more clearly that the number one (1) in the mask marks the limit between the network part and the host, so the network address is (in binary): All the binary numbers to the right of the network part in the mask are filled with zeros. Let us now turn the above network address into decimal. We will get: à This implies that the IP address belongs to the network Now the router can decide to which network to send our IP packet. 1.3 Transport protocols Note: We should not confuse these protocols with voice transport protocols RTP. RTP is a more specialized protocol that even can be mounted on UDP (A protocol that will be explained here). The transport protocols covered here are for general purposes and rest directly on top of the IP protocol. TCP protocol As we had said, the IP protocol does not guarantee that data will reach its destination. It only makes the best effort to ensure that it will arrive. Due to this best effort, a protocol was needed to be responsible for the data transmission. The Transmission Control Protocol or simply TCP was designed for this reason. This transport protocol is transmitted on top of the IP protocol.

7 TCP helps by making sure that the transmitted data is free of errors and is received by the applications in the same order it was sent. If data is lost in the way, TCP introduces a mechanism so that this data can be resent. Obviously, this implies an extra load of information on the data flow since there is additional control information to be sent. That is why TCP is a good protocol for session control but not so good for real-time information transmissions like voice. However, TCP plays an important role in many protocols related to the Elastix server. TCP introduces the concept of a port that is nothing else but an abstraction to be able to relate data flows with specific network services (or higher-level protocols). For example, port 80 is associated with the Web service or HTTP protocol, while port 25 is associated with service or SMTP protocol. UDP protocol UDP (User Datagram Protocol) is another transport protocol. It is different from TCP, as this protocol does not care if the data arrives with or without errors and does not care if it arrives in sequence. The question is then, where and why is this protocol used? Well, a transport protocol does not necessarily have to guarantee that the information reaches its destiny or that it arrives in sequence. This is just an extra feature. It is somewhat like a merchandise transport service. Let us imagine a fleet of motorcycles that offers transport services in a city and let us imagine that that fleet is hired for transporting a great shipment of files from one office to the other. The company will be in charge of dividing our shipment of files and distributing it into quantities or packets that can be transported in each vehicle. The fleet will do whatever it needs to make sure the shipment reaches its destination. This is in essence what a transport service does. However, we can choose two kinds of services: one that guarantees that the merchandise arrives safely and one that does not. These two services are analogous to the two kinds of transport protocols that we are describing here: UDP and TCP. In the end, UDP divides the data into packets, also called datagrams, to be transported inside the IP packets to its destination. Since it is not necessary to include too much control information, the UDP protocol reduces the quantity of information necessary in the packets, which makes it a faster protocol than TCP. This makes it suitable for the transmission of information that must be sent in real time, such as voice. This is why VoIP voice applications are transmitted with this protocol.

8 1.4 Voice codification It is clear that to transport voice there are some protocols used such as SIP, IAX and others such as RTP or RTCP. But voice is an analog wave that needs to be transformed into a digital format before being transmitted. Logically we could try to transmit it just as it results from an analog to digital conversion (ADC) but we are in a packet network so we need to packetize this information. There also exists the possibility of compressing the result of ADC conversion with an algorithm so not to waste network resources. The search for an optimal compression format generated some alternative transmission formats called codecs. Note: Codecs are not really exclusive of VoIP since they are also used in other kinds of digital communications. Codecs The word codec comes from abbreviating the words codification and decodification. Its main purpose is to adapt the digital information of the voice to obtain some benefit. This benefit in many cases is the compression of the voice in such a way that we can use less bandwidth than was necessary. Some of the codecs supported by Asterisk and commonly used in VoIP communications are G.711, G.722, G.729, GSM, and ilbc among others. Let us briefly look at four of them: G.711, G.722, G.729 and GSM. G.711 G.711 is one of the most used codecs of all time and comes from ITU-T standard that was released in It comes in two varieties called u-law and a-law. The first version is used in the United States and the second is used in Europe. One of its characteristics is its voice quality since it hardly compresses it. It uses an 8 bit - 8kHz sampling, resulting in 64kbit/s. It is the recommended codec for LAN networks, but you should think twice about using it for remote links due to the large amount of bandwidth that it consumes. The support for this codec is a default setting in Elastix.

9 G.722 G.722 offers a sampling rate of 14 bits at 16 khz, much higher than G.711. In other words, it recovers more data from the audio signal, which results in a superior voice quality. In fact, the codec is used in many VoIP telephones marked as high definition (HD.) Something very interesting about G.722 is that despite having a sampling rate that is higher than G.711's and the fact that the resolution of each sample is also higher than G711, the bandwidth consumed is almost the same, 64kbit/s. This is possible due to G.722 codec using a different algorithm. If we look quickly at the human voice bandwidth that this codec can capture (according to Nyquist's theorem seen in the last chapter), we will see that the sampling rate of 16kHz allows us to digitize frequencies of up to 8kHz. That is, twice the bandwidth of G.711, including high frequencies. These high frequencies contain high tones that make the voice clearer and more understandable. With the recent arrival in the market of phones that claim to have High Definition Audio (HD), we have seen an increase in the use of this codec. The reader may think that this is a new technology; however, G.722 is a 1988 standard. G.729 This is also an ITU recommendation whose implementation has been historically licensed. This means that usually you have to pay for them. The advantage in using G.729 lies mainly in its high compression and the resulting low consumption of bandwidth. This makes it attractive for Internet communications. Despite its high compression, G.729 does not deteriorate voice quality significantly and for this it reason has been widely used over the years by many manufacturers of VoIP products. G.729 uses 8kbit/s on each channel. If we compare this to the value in G.711 (64kbit/s) we'll see that it consumes 8 times less bandwidth, which at a glance if a significant saver of our resources. Note: There exists a variation of G.729 that use 6.4kbit/s and 11.8kbit/s. To use G.729 channels in Elastix a license must be acquired for each channel.

10 GSM Many people correctly assume that the GSM codec has something to do with cell phone communication standards. The standard that defines cell-phone technology GSM (Global System formobile communications) includes this codec. The advantage of this codec is also its compression. However, the voice quality is deteriorated compared to other codecs. You will notice this when you talk through a GSM cell phone at some point. GSM compresses approximately at 13kbit/s and comes ready to be used by default in Elastix. 1.5 Protocol overload By what we have seen above, transporting voice from one place to the other in a packet network requires the help of some protocols. In addition, we have noticed that these protocols transmit additional data that uses up bandwidth apart from the actual voice. Some of them are Ethernet, IP, UDP, and RTP. All this makes the real bandwidth required to transmit voice to be higher than the actual codec. For example, to transmit voice using G.711, in theory we should be using 64Kbps (according to the codec) when in reality we are using 95.2Kbps of bandwidth. In other codecs with more compression, the overload is even more significant (percentage-wise.) Let us do a simple exercise using as an example the G.711 codec. Let us see the following figure to illustrate the exercise better: Illustration of protocol overcharge using G.711

11 Let us calculate the bandwidth for G.711 basing ourselves in the above figure, which shows us the waste or overhead of each protocol. Let us add up all of these sent bytes. Bytes transmitted for each 20ms of audio = = 238 bytes Bits transmitted for each 20ms of audio = 238 bytes * 8 bits/byte = 1904 bits Bits transmitted each second = 1904 bits/20ms * 1000 ms/sec. = 95,200 bits/s = 95.2Kbps! Comparing codecs Below we show a table that shows the overhead for some of the most popular codecs supported by Asterisk. Codec Codec Bandwidth Real Ethernet Bandwidth Overhead Percentage G Kbps 95.2 Kbps 48.75% ilbc 15.2 Kbps 46.4 Kbps % G.729A 8 Kbps 39.2 Kbps 390%

12 2.1 What is Asterisk? Asterisk Essentials Asterisk is telephone exchange (PBX) software with VoIP capability that is distributed with free licensing. Parting from this very basic concept, Asterisk is not just any PBX; it is a PBX rich in features that before were only accessible by purchasing expensive products. This has resulted in many businesses considering Asterisk as a serious option when planning their telephone project and, for this reason; Asterisk has been very well received around the world. Seeing this business opportunity, many manufacturers have started to offer telephone hardware that is compatible with Asterisk, mostly electronic cards for PSTN connection. This has caused Asterisk-based PBX s offerings to rapidly grow in recent times. Asterisk is one of the most important components in Elastix and it provides most of the distribution s telephone features. 2.2 A Brief History of Asterisk Asterisk was conceived and developed by Mark Spencer. At first, it was because of a personal need, since he needed a PBX for the small support business he was founding, Linux Support Services. At first, he thought about acquiring a PBX but he soon realized that it was far too much for his budget. So, after some thought, he decided to create his own and started to code what we now know as Asterisk. In 1999, when he had a functioning program, he decided to make it free under a GPL license. One may not think it, but Mark was no newcomer to the world of open-source software. He had already participated actively in the development of the messaging client Gaim, now known as Pidgin, one of the most popular pieces of open-source software. In 2002, Linux Support Services turned into Digium, which today offers Asterisk-related products and services and handles the development of this software. We should point out that right at the beginning Mark realized that his software needed to interact with telephone hardware. That is when he discovered the Zaptel project, an opensource project created by Jim Dixon. Zaptel had the aim of creating open drivers for computer

13 telephone cards. From then on, Asterisk and Zaptel would walk hand in hand to the point where today both services are maintained by Digium. In 2008, Digium decided to rename the project they maintained with Zaptel to DAHDI. 2.3 Functionality Provided by Asterisk As we have said before, Asterisk is a PBX software that is rich in features. While writing this book, the most relevant supported features are listed below: Automatic call answering Call transference Do not disturb option Call parking Call pickup Call monitoring and recording Voic Conferencing Call reporting Waiting lines Call waiting Caller ID Caller ID blocking Outgoing and incoming fax Interactive extensions directory listing Interactive voice response (IVR) Music on Hold Behavior management by schedule Follow me Automatic call answering Asterisk provides the ease in configuring an automatic answering service that helps manage the flow of incoming calls by answering them automatically with no need for a live operator. This feature has been designed for businesses that receive a high call flow and that do not possess the necessary staff to manage the traffic of incoming calls. The answering service responds to the digits that are entered into the number pad by the person calling, sending the calls to specific extensions, providing access to pre-recorded information and can take messages 24 hours per day.

14 Call transfer There are two methods in Asterisk to transfer calls: attended transfer and blind transfer. In attended transfer the person at the destination is called first to notify him that a call will be transferred to him. In blind transfer, the call is simply transferred without previously speaking with the person at the destination. Do not disturb option This option allows the user to configure the extension so that it will not receive calls for a period. Any incoming calls to that extension will be automatically rerouted to that same extension s voice mail. Call parking Call parking allows the user that receives a call to send it to a parking space, so he can take it again from another extension. This kind of action is useful in many situations, like when a technician receives a call at his desk that requires him to be in front of a server that is three stories above. In this case, the option is to send the call to a virtual room or parking space where the caller will listen to the configured music on hold. The technician can retake the call once he arrives at the server. Call pickup This option allows one to take a call that is ringing at an extension that is not one s own. Call monitoring and recording Asterisk allows the monitoring and recording of calls. Monitoring is usually used in quality control for telephone operators and sales agents. For this, there is the feature in which by typing in a predetermined code, one is able to listen in on a conversation taking place on any one extension. Additionally, it allows the recording of conversations from any extension randomly or previously programmed. These recordings will be stored in the Asterisk server s hard drive for later revision. Voic Voic is an application that allows messages left by callers that were not answered to be stored and listened to. To access the voic inbox you must enter a predetermined password. Asterisk will ask for the password and then you will have access to the available messages.

15 Conferencing A conference is nothing more than a phone call in which several participants can speak at the same time. To access this feature, conference rooms are created. Each room is associated with a number, as if it were an extension. Then, to access a conference all you have to do is dial the corresponding number for that conference room. It is possible to configure some interesting settings for a determined conference room, for example limiting the number of speakers, asking for a password, and moderating the conference room. Call reporting Asterisk generates Call Detail Records (CDRs) and can store them in a database. By accessing this database, reports can be generated that detail what numbers are dialed by each extension, whether a call was answered or not, how long the call lasted, or through what port or phone line the call was made Queues This feature allows a great number of callers to stay on hold until a representative or resource is available to assist them. This ensures that callers will not go automatically to voic , which can ensure that a caller always has the opportunity of being assisted by a person. The time the user stays on hold can be used to let him know of the company s services or products, special offers, informative messages or there can simply be music on hold. Call waiting This feature allows the person that receives a call while being on another call to temporarily interrupt the current conversation to answer the incoming call. To alert that person that a second call is coming in, there is a characteristic sound. In some scenarios this sound is not desirable, for example in telephone connections for data transfer (i.e. fax.) In these cases, you can choose to turn this feature off, since it creates fax transmission errors. Caller ID This signal is sent in between ring signals, or during the call establishment process, before it is answered. Asterisk takes advantage of this and completely supports its use at an IP extensions level. However, at the level of public commutated phone networks, the provider must turn on or provide this feature.

16 Caller ID blocking This feature prevents someone with caller ID to see the number from which you are calling. This caller ID hiding feature can be total or selective. Not all public telephone networks support this feature. In Asterisk, this feature also includes blocking an incoming call according to the ID. Incoming and outgoing fax Asterisk allows automatic detection of when a caller is trying to send a fax. As a part of the welcome message for the PBX, you can add a phrase that says, If you want to send a fax, send it now. This allows us to eliminate paper, to do without machines or telephone ports exclusively dedicated to fax and to digitize the information received as fax. After being digitized, this document can be ed to a specific account for review. Interactive listing of the extensions directory Asterisk has the telephone directory for all of the company staff (first name, last name, and extension) in its database. This feature allows the person calling to enter the first four letters of the first or last name of the person with whom he wants to speak. This can give callers an alternative if they do not know the extension of the person they are trying to reach and only know their first or last name. Interactive voice response (IVR) Through this feature, access is provided to telephonic options that improve the way a telephone system receives and distributes calls. With an IVR menu, it is possible to distribute incoming calls in a precise way. This will help raise employee performance and customer satisfaction, since it speeds up access to the services and data possessed by the company. With this feature, automatic processes can be put into place that in certain cases can even allow working without a receptionist. With the use of this feature, you can create number or voice command activated menus to even look up data in other computer systems or databases. Music on hold Asterisk allows the use of any mp3 or wav file as background music. This will be played when an internal or external caller is waiting to be attended by a representative of the company. These files can even be grouped by categories to produce different melodies for different caller groups.

17 Behavior management by schedule It is possible to configure Asterisk so that it will have a different treatment for calls (both incoming and outgoing) depending on the time of day, day of the month, or month of the year. The most basic example is when you want the incoming calls received outside of office hours to receive a message suggesting them to call again the next day. Follow me This feature, allows to predefine a sequence of simultaneous or sequenced calls to locate a user. For example, you could define that when the company manager s extension rings and is not answered the call should be rerouted to his cell phone and other contact numbers. You can specify grace periods before the next number is dialed. 2.4 Asterisk Operation Asterisk directories Asterisk organizes its files by distributing them among several directories. The most important of these are the following: Directory /etc/asterisk/ /usr/lib/asterisk/modules/ /usr/sbin/ /var/lib/asterisk/agi-bin/ /var/lib/asterisk/mohmp3 /var/lib/asterisk/sounds /var/spool/asterisk/ /var/run/ /var/log/asterisk/ Description Where the Asterisk configuration files reside Contains Asterisk s modules Includes Asterisk s binary Where the AGI scripts reside Contains the Music On Hold mp3 files Includes the sounds that Asterisk uses as voice prompts Where Asterisk saves the files generated by its operation, such as voic s and call recordings Saves files with process information (PIDs) Where the Asterisk log files reside, such as CDRs or /var/log/asterisk/full

18 Modular structure Asterisk uses the idea of modules to extend its functionality. The functionality residing in the modules is so important that without them Asterisk alone would not perform any interesting functions. These modules are in the folder /usr/lib/asterisk/modules/ and are files with the.so extension Modules can be controlled through the configuration file /etc/asterisk/modules.conf. This file tells Asterisk whether to load a specific module through the directives load andnoload. By default in Elastix, all modules try to load. The following is a fragment of the file modules.conf that comes by default with Elastix. Here you can observe how it tells Asterisk not to load some modules during startup. ; ; DON'T load the chan_modem.so, as they are obsolete in * 1.2 noload => chan_modem.so noload => chan_modem_aopen.so noload => chan_modem_bestdata.so noload => chan_modem_i4l.so ; Ensure that format_* modules are loaded before res_musiconhold ;load => format_ogg_vorbis.so load => format_wav.so load => format_pcm.so load => format_au.so Additionally, you can load modules in real time through the Asterisk console (not to confuse this with the Linux terminal) with the command: module load <module> Further ahead we will see some more about the Asterisk console or CLI.

19 2.5 The DAHDI Project Even though it is a great piece of software, Asterisk cannot do all the work by itself. One of the software packages that complement it is called DAHDI. DAHDI is a group of drivers to control telephonic hardware like PCI or PCI Express cards, which allow us to connect to the PSTN. These drivers communicate with Asterisk through an Asterisk module called chan_dahdi.so that is configured through the configuration file /etc/asterisk/chan_dahdi.conf. Additionally DAHDI has its own configuration files, independent from Asterisk, which are found in the folder /etc/dahdi/. Note: In versions of Elastix before 1.5, the software from project Zaptel is used instead of DAHDI. This means that the configuration files for the telephony drivers have different names. In these cases, it is recommended, as reference, to consult the first edition of this book. A Brief History of DAHDI Simplified diagram of the interaction between Asterisk and DAHDI Regardless of the name change from Zaptel to DAHDI and all the confusion that this generated at the beginning, we cannot forget that DAHDI s origin was in the Zaptel project. First, because the people working on the original project deserve their due credit, second because we should not forget history. Following is a brief summary of that history.

20 In the year 2000, Jim Dixon started project Zaptel, whose name was an abbreviation of the Zapata Telephony Project. He named it in honor of Emiliano Zapata, hero of the Mexican revolution. The drivers were launched with a GPL license so that anyone could have access to the code. Initially Zaptel had the drivers for the Tormenta (Spanish word for Storm) cards, made by Zapata Telephony. It was not long before Digium started to improve the drivers and extended support for new hardware models, among them those manufactured by Digium itself. That is how Digium became Zaptel s main developer. After a few years of maintaining Zaptel, Digium realized that Zaptel was a registered brand by a card selling company and to avoid any future confusion, decided to change the name of its drivers. The name chosen was DAHDI, which is an acronym for Digium Asterisk Hardware Device Interface. Digium ended official Zaptel support after Asterisk version Asterisk Configuration Configuration files Asterisk can be configured through some configuration files found on the path /etc/asterisk. There are dozens of configuration files in this directory and they are in plain text to make their modification easier. Due to this, we can use the vim editor or the command line editor of our preference to edit them. Even though all files are important, not all of them are necessary. Elastix has already preconfigured some of them to prevent further modification. The most important are explained in the following table: File extensions.conf sip.conf iax.conf chan_dahdi.conf Description Defines the dialing plan. In Elastix, there are other files to better organize the dialing plan. These additional files start with the termextensions_ Configures SIP devices Configures IAX devices Defines configuration for DAHDI-type channels

21 Comments on the Configuration Files Asterisk configuration files can contain comments. A comment starts with a semicolon. Once this character is written, the rest of the line turns into a comment. That is, this line is ignored by Asterisk when reading the file. Let us see an example of a comment. ; ; The following is a comment block. ; As you can see, I can write anything ; in here because it will be ignored by Asterisk. ; Comments are very useful for writing ; notes and information relative to the configuration ; we are writing. ; Now I will continue with the directives... ; 2.7 Dial plan #include sip_general_custom.conf #include sip_nat.conf #include sip_registrations_custom.conf #include sip_registrations.conf The dial plan in Elastix is mainly in the file extensions.conf. From this, one there are other important files included which also contain portions of the dial plan. The most important are: extensions_additional.conf: Here is where the variable plan written for FreePBX resides, an important component of Elastix. If changes are made manually to this file, these changes will be lost when they are overwritten by FreePBX. extensions_custom.conf: Here the user can manually define a dial plan. This file is not touched by FreePBX so it is safe to edit it without fear of losing any changes. All of the dial plan files are in plain text in a format defined by Asterisk. The content of these files is divided into sections called contexts. Contexts Contexts group the logic of the dial plan. Contexts also can include other context with the clause include or redirect the flow of execution of the dial plan to other contexts depending on the conditions. For example, after examining the prefix of the dialed telephone number, a context can contain the necessary logic to route the call to the corresponding main line. A context can group the logic necessary for the application of an alarm clock. It can also include the logic to have access to voic or to authenticate a PIN number. In general, practically

22 the whole dial plan is grouped into contexts. Contexts are the groupers and organizers of the dial plan. Each context must have a unique name and this name is between the symbols [ and ]. You can observe two contexts called app-pickup and ext-test in the example below: [app-pickup] include => app-pickup-custom exten => _**.,1,Noop(Attempt to Pickup ${EXTEN:2}) exten => _**.,n,pickup(${exten:2}) [ext-test] include => ext-test-custom exten => 7777,1,Goto(from-pstn,s,1) exten => 666,1,Goto(ext-fax,in_fax,1) exten => h,1,macro(hangupcall,) Fragment of the file extensions_additional.conf to illustrate contexts. Reserved contexts There are two context names that are reserved for a special purpose which are [general] and [globals]. Extensions Contexts contain instructions, known as extensions. These extensions are lines from the dial plan that have the following syntax: exten => extension_name, priority, application Note: We usually associate the word extension with phone numbers, but in Asterisk, the term extension goes beyond what we normally know. An extension is a logic series of the dial plan. What s more, when we dial a number from our telephone it does not necessarily have to ring another telephone since we can create the logic we want for that given number. For example, we can simply make the call hang up, to reproduce a given message or functions that are more complex. A typical extension looks like the following: exten => 4567,1,Answer()

23 Extension name The extension name is a numeric identifier for the extension. Let us call it telephone number for now to simplify the matter. There is a special non-numerical extension called extension s, to which we will give special attention. Extension s means any extension. The letter s comes from the word start. An example of the use of extension s is the following: [test] exten => s,1,answer() exten => s,2,noop( Test ) Extension s is used when the user that made the call has not pressed the extension number. For example, this is the case for an incoming call still listening to the IVR welcome. Priority Priority indicates the order in which an extension is executed inside a context. It is necessary to name the first priority 1 so that when Asterisk reads a context the first thing it looks for is this priority. There also exists a special priority, which is priority n. Priority n means that the priority of the previous instruction is added to one. This is very useful for making easier the writing of the dial plan. For example, let us check the following dial plan: exten => s,1,noop( Hello, this is just an example ) exten => s,2,noop( to demonstrate ) exten => s,3,noop( how to use priority n ) This dial plan could be written in the following manner using priority n: exten => s,1,noop( Hello, this is just an example ) exten => s,n,noop( to demonstrate ) exten => s,n,noop( how to use priority n ) Priorities can also contain tags to make their identifying easier since if we always use priority n it will be difficult to access a determined location determined inside the context. This can be done with the syntax n(tag_name).

24 Application It is in applications where the real power resides of the Asterisk dial plan. To make an analogy, the applications are the equivalent to the functions in a programming language. The applications will allow us to answer or hang up a call, play waiting music, and jump to other contexts, among many other functions. Applications can also receive parameters. For example, the application Dial( ) needs you to indicate which number to dial so that it can do its work. Further ahead in the present article, we will explain in detail the most common applications in the dial plan. Variables Variables in the dial plan are a similar concept to variables in the a programming language. That is, they are abstractions that can store information of a variable nature. A variable has the following syntax: ${NAME} Here NAME is the name of the variable. Variables can be of three kinds: Global, Channel, and Predefined. Global They are the variables defined in the [globals] section of the file /etc/asterisk/extensions.conf. The word global means that they can be referenced from anywhere. Channel They are the variables that can be defined using the command (or application) Set( ). Predefined Asterisk administrates some variable names on its own. These names are predefined and contain certain information depending on the behavior of the call and other factors. Some predefined variables are: ${CALLERID(all)} ${CALLERID(name)}

25 ${CALLERID(num)} ${CHANNEL} ${CONTEXT} ${EPOCH} ${EXTEN} ${SIPUSERAGENT} ${UNIQUEID} Note: There is an additional kind of variable introduced in Asterisk 1.6 called a shared variable. This variable allows two or more channels to have access to what would otherwise be a channel variable. Below we reproduce a segment of the file extensions.conf that is distributed by default with Elastix. Here we can see some predefined variables like ${UNIQUEID}, ${BLINDTRANSFER} and ${EPOCH}, and channel variables like ${CALLFILENAME}. [macro-record-enable] exten => s,1,gotoif($[${len(${blindtransfer})} > 0]?2:4) exten => s,2,resetcdr(w) exten => s,3,stopmonitor() exten => s,4,agi(recordingcheck,${strftime(${epoch},,%y%m%d- %H%M%S)},${UNIQUEID}) exten => s,5,noop(no recording needed) exten => s,999,mixmonitor(${callfilename}.wav) Most common applications Answer Syntax: Answer([delay]) Answers a channel immediately if it s ringing. It can optionally receive a delay parameter that tells Asterisk that it must wait a determined number of milliseconds after answering the call to go to the next application. Background Syntax: Background(filename1[&filename2...][ options[ langoverride] [ context]]) Plays a list of audio files in the background and returns control to Asterisk, which can continue to execute the dial plan while the audio is played. When it s done playing the last file, it s

26 execution ends so if you want it to keep waiting for an extension you should use the application WaitExten() with it. This application is typically used to reproduce the welcome message of an IVR. The Background application can receive some options: Option s n m Description Skip the audio playing if the channel isn t answered Don t answer the channel before playing the files Stop playing audio if a dialed digit coincides with an existing extension in the destination context. Playback Syntax: Playback(filename1[&filename2...][,options]) Plays one or more audio files. The difference with the Background application is that Playback plays the whole file until the end and does not return control until it finished playing. Note: It is not necessary to specify the audio file s extension in this function. The audio files are searched for in the directory /var/lib/asterisk/sounds Hang-up Syntax: Hang-up() Hangs up the channel and returns the -1 value. Goto Syntax: Goto([[context ]extension ]priority) Skips the execution of the dial plan and goes to a defined context, extension and priority. If there is only one parameter, it is understood that it is a priority inside the same context. Dial Syntax: Dial(type/identifier, timeout, options, URL) This command is responsible for dialing a specified channel and links it to the current channel.

27 2.8 Asterisk CLI Asterisk CLI is the name for the Asterisk console. That is, the command line interface to control Asterisk directly. To enter the CLI we must execute the following command from the Linux terminal: # asterisk -r Asterisk , Copyright (C) Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk currently running on elastix (pid = 17080) Verbosity is at least 3 elastix*cli> As we can see, we get a CLI> prompt from which we can execute a series of commands that we will go into briefly. CLI can also give us real time information about Asterisk s activity. In addition, we can control the degree of detail with which we want to see that information with some commands. The most used ones are the following: elastix*cli> core set debug 9 Core debug was 3 and is now 9 elastix*cli> core set verbose 9 Verbosity is at least 9 The higher the levels of the variables debug and verbose, the more detailed the displayed information. By default, the level of debug is 0 and for verbose it is 3 in Elastix. Note: Due to the great deal of information that can be generated with these commands, it is important to reestablish the values for the debug and verbose variables their minimum when we are done with our diagnostic tasks. Otherwise, Asterisk will continue to generate this detailed information when we do not really need it, which is extra work that consumes system resources. In systems with a lot of activity, this additional consumption can be significant and can affect the quality of the calls.

28 CLI Commands Below we provide a brief list of the most common CLI commands. A complete command list can be found in Appendix B. Command agi set debug on core set debug channel core set debug core set verbose Description Allows the debug of AGI scripts. It s turned off with agi set debug off Allows debug in a specific channel Allows changing of the debugging level. It s turned with core set debug off Raises the intensity of logging. Very useful in diagnostic. core show channels Shows updated information in active channels. dialplan reload dialplan show help iax2 set debug iax2 show peers iax2 show registry module reload pri set debug on core restart gracefully core restart now core restart when convenient sip set debug ip Reloads the whole dial plan Shows the dial plan Shows a listing of all available CLI commands Starts debug at the IAX level. Is turned off with the command iax2 set debug off Shows the defined peers and some additional information like the IP address from which they re connecting, among other data. Shows information from the IAX registry and other information like the status and IP address to which you are connecting. Reloads all Asterisk modules. Is useful to apply changes made to the configuration files. Starts debug of the PRI interface spans. It turned off with pri set debug off Reboots Asterisk in an ordered manner. That is, it doesn t allow new calls to start and it waits for existing call to end before proceeding with the restart. Reboots Asterisk immediately Restarts Asterisk when there is no activity in the Central. Adequate for production environments in which we have no other option but to restart central but don t want to cut existing calls short. Sets SIP debug at the level of a specific IP address

29 sip set debug peer sip show peers sip show registry core stop gracefully core stop now core stop when convenient dahdi show status dahdi show channels Sets SIP debug at the level of a specific peer Shows a list of all configured SIP peers and information for each one, like their status. Shows a list with the SIP registry and relevant information for each one. Stops Asterisk in an ordered manner. That is, it does not allow new calls and it waits for existing calls to end before stopping Asterisk. Stops Asterisk immediately Stops asterisk when there is no activity in the central. It is adequate for production environments in which we have no other option but to turn off central but do not want to cut existing calls short. Shows a listing of DAHDI channels and a report of alarms and errors Shows a listing of DAHDI channels and some operating parameters elastix*cli> dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-pstn default 2 from-pstn default 3 from-pstn default 4 from-pstn default 5 from-pstn default 6 from-pstn default 7 from-pstn default 8 from-pstn default elastix*cli> DAHDI channel listing, result of the command dahdi show channels

30 Asterisk File structure 3.1 File Structure Asterisk is a complex system, composed of many resources. These resources make use of the filesystem in several ways. Since Linux is so flexible in this regard, it is helpful to understand what data is being stored, so that you can understand where you are likely to find a particular bit of stored data (such as voic messages or log files). Configuration Files The Asterisk configuration files include extensions.conf, sip.conf, modules.conf, and dozens of other files that define parameters for the various channels, resources, modules, and functions that may be in use. These files will be found in /etc/asterisk. You will be working in this folder a lot as you configure and administer your Asterisk system. Modules Asterisk modules are usually installed to the /usr/lib/asterisk/modules folder. You will not normally have to interact with this folder; however, it will be occasionally useful to know where the modules are located. For example, if you upgrade Asterisk and select different modules during the menuselect phase of the install, the old (incompatible) modules from the previous Asterisk version will not be deleted, and you will get a warning from the install script. Those old files will need to be deleted from the modules folder. This can be done either manually or with the uninstall make (make uninstall) target. The Resource Library There are several resources that require external data sources. For example, music on hold (MOH) can t happen unless you have some music to play. System prompts also need to be stored somewhere on the hard drive. The /var/lib/asterisk folder is where system prompts, AGI scripts, music on hold, and other resource files are stored. The Spool The spool is where Linux stores files that are going to change frequently, or will be processed by other processes at a later time. For example, under Linux print jobs and pending s are normally written to the spool until they are processed. For Asterisk, the spool is used to store transient items such as voice messages, call recordings, call files, and so forth. The Asterisk spool will be found under the /var/spool/asterisk directory. Logging Asterisk is capable of generating several different kinds of log files. The /var/log/asterisk folder is where things such as call detail records (CDRs), channel events from CEL, debug logs, queue logs, messages, errors, and other output are written.

31 Elastix Overview Elastix is an Open Source Software to establish Unified Communications. About this concept, Elastix goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution. Unified Communications With Elastix The Elastix project begun as a call report interface for Asterisk and was released in March of Later that year the project evolved into an Asterisk Based distro. Telephony was the traditional way that lead communications the last century, that s why many Companies and users focus their requirements on their necessities to establish telephony communications on their organizations, and confuse unified communications distros with a telephone exchange system. Elastix, not only provides telephony, it integrates other communication alternatives to make your organization environment more productive and efficient. Elastix include the following communication media: Features There are new ways of communication every day and the addition of features and functionality must be constant. Elastix it s capable of establish an efficient environment on your organization with the addition of many features that allows to integrate other locations of your company to centralize your business and take it global. A user in your corporation located in South America shares the same functionalities of another located in Asia besides having direct internal communication. Some of the basic Features of Elastix include: Voic Fax-to- Support for softphones Web Interface Configuration Virtual conference rooms Call recording Least Cost Routing Extension Roaming PBX Interconnection Caller ID CRM Advance Reports

32 Installing Elastix Before beginning, make sure you have a copy of the latest stable version of Elastix (the present work is based on version 2.0.) This can be downloaded from the download page of the Elastix Project at Choose the appropriate version depending on whether your machine has a 32 or a 64-bit processor. Loading from CD Elastix is distributed as an ISO image file, which can be burned to a CD with any CD recording software. Once this image is burned on the CD, insert it into your drive and turn your computer on. Make sure that your computer starts up from the CD-ROM drive otherwise, you need to set this option from the BIOS. If everything goes alright, you should get a screen like this one: Initial installation screen Note: The Elastix installation CD will format ALL of the hard drive during the installation process so make sure that you do not have any data that you will need on your hard drive. The installation process will start automatically after ten seconds. You can press the Enter key if you do not wish to wait that long.

33 Preferences and welcome After loading some programs necessary for the installation, the process will ask us to enter the language for the installation. Choose the language using the arrow keys on your keyboard. Using TAB, move the cursor on OK and press the Enter key. Language selection for the installation After that, go on to choose your keyboard type according to its layout. If your keyboard s layout is from the United States, select the us option. Keyboard-type selection

34 Hard disk partitions The following is the critical step in the installation since it will configure the hard drive partitions. If you are an experienced administrator, you can choose Use a custom design. Otherwise, choose Remove all partitions on selected drives and create default layout. In this example, we will select the latter option. Hard drive partitioning Partitioning Confirmation Now the installation process will ask us if we want to review and modify the partitioning layout. Select the No option and press the Enter key to continue. Review/Modify partitioning layout

35 Network parameter configuration Now it is time for the network parameter configuration. The installation program will ask if you wish to configure these parameters in the network cards of your server. Select Yes. Network configuration For each network card installed, it will ask if you wish to activate it during startup and which IP support you will activate in it. The IP protocol we recommend you choose support for is IPv4. Press OK. Parameter activation on the network card eth0 If you have a Dynamic Host Configuration Protocol (DHCP) server in your network for the dynamic assigning of IP addresses, select the option Dynamic IP configuration. Otherwise, you will have to assign an IP address manually selecting the option Manual TCP/IP configuration.

36 IP address configuration and network mask The next screen will ask us to enter additional network parameters such as the gateway, primary DNS and secondary DNS. Gateway and DNS configuration Now we need to name our server with a hostname. It must be taken into account that a correct host name has the form name.domain. Your network administrator will know what host name to use. If you do not have one, enter a fake name since later it is possible to change it. Choose OK to continue. Hostname configuration

37 Time zone, root password and packages In the next screen choose the time zone for your region and press OK to continue. Time zone selection We are now going to enter a series of high-level passwords for your Elastix system. It is strongly recommended that you clearly note down each password, especially as you should be using complex passwords. Do not take this suggestion lightly as even seasoned installers of Elastix have had to re-install Elastix because they failed to write these important passwords down. It is time to specify the root password that will be used to access the linux prompt via the console or SSH. Remember that this password is a critical part of the system s security. Due to the importance of this step, you have to confirm the password by entering it again. Press OK to continue with the installation. Specifying the administrator password (root)

38 Formatting and file copying From here, the installer program will partition and format the server s hard drive and all the data stored previously on the hard drive will be lost forever. In addition, all of the files necessary to run Elastix will be copied onto the hard drive. File copying to the hard drive Once the package copying to the hard drive is finished, the server will restart automatically. At this point, the server will eject the CD, please remove it. If you don t it will try to boot off of the CD and not the hard drive where we now want to boot from. Automatic restart and various passwords Let the system start up by itself while making sure not to press any keys at the next screen. Elastix startup screen

39 After loading the programs needed for its operation, Elastix will ask you for passwords for the important components that make up your system. This password entry phase will only happen on the first startup after installation. The first password that it asks for is for the MySQL database administrator. Enter a password and continue with the OK button. MySQL administration password You will have to confirm the password previously entered for MySQL administration in the next screen. MySQL administration password confirmation The next password will be used to enter with administrator privileges into the Elastix graphic interface. In addition, the same password will be used to enter the FreePBX, VTiger and A2Biliing system components. Once the password is entered, advance to the next screen by pressing OK.

40 Elastix password and its components As in the last case, you will need to confirm the password previously entered for Elastix and its components. After entering it once more, press OK to continue. Confirming the password for Elastix and its components

41 Ending the installation The installation process will end when you get an invitation screen to enter the main server console, like is shown in the following figure: Authentication for entering the Elastix console To access the Elastix console, simply enter root at the pbx login prompt. The password is the one you chose during the installation process. Course content Introduction to VoIP Asterisk as a VoIP server Asterisk Essentials Asterisk File structure Elastix Overview Elastix Basic Setup Installing Elastix IP telephone configuration Hardware configuration Elastix Advance Setup Make your first call