- 3CX Phone System
- IP Session Board Controllers
- VoIP Gateways
- VoIP IADs
- VoIP Routers
- Media Gateways
- Line Extender
- FXS gateways connect internal analog devices such as fax machines and phones, and register them as extensions / fax extensions to your 3CX. Those extensions can be addressed as normal phones however their functionality is limited due to the capabilities of the analog technology.
Prepare the Patton FXS Gateway
In order to identify your FXS gateway IP address follow our guide “How to Use the Patton Discovery Tool”. Once the device is detected, download and upgrade the firmware of your Patton FXS gateway. Note: All templates are designed to work with Patton SmartWare firmware and the device should run on the “latest 6.X firmware”. Trinity Firmware/Devices are currently NOT supported.
To update the firmware:
- Open your web-browser to your FXS gateway IP address retrieved in the previous step.
- Login to your account (Default username: administrator, password: leave empty).
- Navigate to “Import/Export” → “Import Firmware” → and in “Download Firmware” choose your firmware .zip file downloaded and then press “Import”.
- After you have successfully uploaded and extracted the firmware press “Reload” to start applying the upgrade.
- It takes some minutes for the firmware upgrade to complete. Do not turn off the power until the update is complete.
Navigate to “FXS/DECT” within the 3CX Management Console and select “Add FXS/DECT”:
- Step 1: Add and Choose Device
- Step 2: FXS Line Setup
- Step 3: Map extensions
- Step 4: Provision the Gateway
- Step 5: Add Fax extension (Optional)
You need to define the device and line specific options:
- Select Brand: Patton
- Select model/device: Patton 2 Ports FXS GW or Patton 4 Ports FXS GW
- MAC Address: Enter the FXS device’s MAC address (underneath the device)
- Press “OK” to create the device.
Navigate to “Extension” tab in the “FXS/Dect” device to assign which extensions should be use from the device.
- Gateway Hostname or IP: Enter your Patton FXS device’s IP address. (the IP address will be set to static in case DHCP is being used)
- Gateway SIP Port: Enter 5060
- Subnet Mask: Enter your Network Mask size (e.g. 255.255.255.0). (if network topologies requires routing informations such as default routes, see here how to add the default route to a Patton device).
- FAX Transmission Mode: In this section select how the Patton gateway should handle fax connections. In case your VoIP provider supports T38 or you have a Patton FXO or ISDN configured for T38 then set the codec to T38. However if you are running a VoIP Provider which does not support the T.38 protocol, select “Fax in Audio”. In T.38 mode the Patton FXS device will actively switch to the T.38 protocol once the call is established.
- Tone Set: Define in which country your are using this device. If the country is not listed request from your telco the used tone set of your line.
Follow the above image in order to define which extensions will be using each port on the FXS gateway. In the example above, port 1 on the FXS Gateway will be mapped to extension 001, port 2 will be mapped to extension 002 and so on. Finalize the configuration by saving the config with “OK”.
Note: If you plan on using fax machines on the FXS gateway, please create dedicated fax extensions in the “Fax Extension” menu (Refer to step 5 for information on how to do this).Reopen the just created FXS device and click on the “Provisioning URL”. A file will be downloaded to your PC which will be used in a few moments.
In case you are using fax machines connected to your FXS gateway, “Fax Extensions” must be used instead of using a regular extension. You can do so by:
- Open your web browser and connect to your Patton FXS device.
- Log in (default username: administrator, no password is required (blank field))
- Navigate to the “Import Configuration” page (e.g. http://192.168.9.163/imp-cfg.html whereby the gateways IP address is 192.168.9.163).
- Upload the configuration file onto the Patton device by using the “Choose File” and “Import” button.
- In order for the configuration to be applied, reload the device. The mapped extension will now be visible in the “Phones” tab (except fax extensions, which will be shown in the “System Extension” tab).
- Go to the 3CX Management Console.
- Navigate to “FAX Extensions → +Add”.
- Define the “Fax Extension”, the “Fax Extension Authentication ID” and the “Fax Extension Authentication Password”.
- After you familiarize yourself with the basic concept of SIP Trunks proceed to add a PSTN gateway into the PBX. All templates are designed to work with Patton SmartWare firmware and the device should run on the “latest 6.X firmware”.
In order to identify your FXS gateway IP address follow our guide “How to Use the Patton Discovery Tool”. Once the device is detected, download and upgrade the firmware of your Patton FXS gateway. Note: All templates are designed to work with Patton SmartWare firmware and the device should run on the “latest 6.X firmware”. Trinity Firmware/Devices are currently NOT supported.
- Step 1: Add and Choose Device
- Step 2: PSTN Line Setup
- Step 3: Port Configuration
- Step 4: Upload Configuration
- Step 5: Add Inbound Rules
- Step 6: Add Outbound Rules
Navigate to “SIP Trunks” within the 3CX Management Console and select “Add Gateway”:
Now you need to define some device and line specific options:
- Select Brand: Patton
- Select model/device: Patton ISDN E1 or Patton ISDN T1 (E1 lines are also known as PRI, S2M, T1 is commonly used in the USA)
- Number of Physical PSTN Ports on device: The amount of PRI ports the device has in total (not how many ports are going to be used).
- Main Trunk No: Define a number of your ISDN lines which should act as catch all destination. Commonly your most important number of all.
- Press “OK” to save the device.
- Registrar: Enter your Patton E1/T1 device’s IP address. (the IP address will be set to static in case DHCP is being used)
- Subnet Mask: Enter your Network Mask size.
- Tone Set: Define in which country your are using this device. If the country is not listed request from your telco the used tone set of your line.
- Number of SIM Calls: Define how many ISDN ports are going to be used. This might be equal to the amount of physical PSTN ports the device has or less.
For each port been used enter the reflecting value below:
For a T1 device
- 1 Port enter 23 SIM Calls
- 2 Ports enter 46 SIM Calls
and so on
For a E1 device
- 1 Port enter 30 SIM Calls
- 2 Ports enter 60 SIM Calls
and so onNavigate to the “Options” tab and in the section “PSTN Gateway Options”. In this section define how calls should be placed to the PSTN network.
By pressing the “Generate config file”, the 3CX Management Console will directly open the web interface of the Patton gateway on the “Import Configuration” page (e.g. http://192.168.9.45/imp-cfg.html). A configuration file is automatically downloaded which then needs to be saved and uploaded onto the Patton device by using the browse and import button.
- Signalling Protocol: Options depend on the interface type. For E1 most commonly is DSS1 whereas for T1, NI2 is the default option.
- Overlap Dialing: This refers to whether or not any further digits are received by the gateway after the initial call is received. Setting this option to “YES” will ensure that all digits will be sent to the 3CX Phone System. This may cost some delaying in the incoming call.
- Hunting Mode: As all PRI ports will be placed in a hunt group, select how the Patton device should allocate the ports while placing outbound calls. Options are: “Normal” (always start from the first port, first channel) or opt to take a “Cyclical” approach (spreading the calls across all ports evenly in a round robin manner). Note: ports that are already in use will be skipped and the next available port will be used.
The device will reboot in order to apply the configuration. After this the “Port/Trunk” status on the Management Console should change from RED to GREEN.To create “Inbound Rules”, click here: Inbound Rules – Routing incoming calls.To create “Outbound Rules”, click here: Outbound Rules – Routing outgoing calls.
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Park Calls0* - Used to park a call. While on a call, click on the Transfer button and dial *0 followed by the parking slot. E.g. to park the call in parking slot 1, dial *01.
Call Park Pick-up*1 - Used to pick up a parked call. E.g. to pick up a call parked in slot 1, dial *11
Call Pick-up*20* - Used to pick up a call which is dialling at another extension. E.g to pick up a call dialling on extension 106, dial *20*106
Enable DND*61 - Used to Enable DND for the extension.
Disable DND*60 - Used to disable DND for the extension.
Change Extension Profile Status*3 - Used to change the status of your profile. *3 should be followed by one of the following to change the profile accordingly:
0 - Available
1 - Away
2 - Out of Office
3 - Available 2
4 - Out of Office 2
Queue Extension Login*62 - Used to log the extension in to the Queues.
Queue Extension Logout*63 - Used to log the extension out of the Queues.
Supervised Call TransferDuring an active call, press TRAN Enter extension number and hit TRAN
Blind Call TransferDuring an active call, Press TRAN Enter number and press ‘OK’ Press TRAN to complete the transfer
Conference CallInitiate call with the first party, once you are connected press the CONF button Dial the number that you would like to include in the conference call Once the second party picks up or answer the call press the CONF button again to join the second caller in the conference call
Retrieving Voicemail by PhoneTo retrieve your voice mail by phone: Dial the voice mail system extension. By default this is 999. If you are outside the company, you can dial this number from any digital receptionist menu. Enter your extension number and when prompted your pin number. Press pound #. A prompt will be played with the number of read and unread messages in your voice mailbox. Press * to play your unread messages. During playback of voice mail, you have these options:
0 - Skip to the next message.
1 - Skip to the previous message.
2 - Repeat current message.
3 - Delete the current message and automatically go to the next message.
4 - Call Back – this will trigger a call to the caller ID of the person who left the message.
5 - Forward message to another extension.
9 - To go to the Voice mail options menu.
# - To hang up the call.
Voicemail Options MenuThe voice mail options menu allows you to set your status, configure your pin code and delete all read messages. It can be reached by entering 9 during message playback or during the welcome prompt. The following options are available:
1 - To set your status.
3 - To dial an external number or internal extension.
4- To delete all messages.
5 - To record your name (used for call by name).
6- To play mailbox information.
7 - To change pin number.
8 - To change your voice mail greeting.
9 - To repeat the prompt.
# - To exit the call.
Send a call to a Voicemail*4 - Used to connect to voicemail of an extension. E.g. to leave a message for extension 106, dial *4106
Hide Caller ID*5 - Dial *5 before the number to be dialled to hide your caller ID from the call.
Billing Code Feature** - This indicates that a billing code is to be used for the call. First dial the number followed by ** followed by the billing code. E.g. if the number to be dialled is 956322 and the billing code is 562, then the full number to be dialled is 956322**562
Emergency CodeThe Emergency code is used to toggle the status of the Phone System between In Office to Out of Office. This code is not defined by default. If you want to use this code, you will need to define your own code from the Dial Codes page.
Create an Ad-hoc Conference Bridge from a Desk PhoneDial the conference extension number, by default this is 700. You will be requested to enter a conference ID. This can be any number, for example ‘100’. If the phone system requires a security PIN to create a call conference, you must enter it after the conference ID, separated by a *, for example: 100*0000 (where 100 is the conference ID and 0000 the system wide conference PIN). If you are the first participant, the conference system will ask you to confirm creation of the conference. Press * to confirm or # to cancel. You will be asked to speak your name after the beep and press a button to continue. You will now enter the conference. If you are the first caller, you will hear music on hold, whilst you wait for the other callers. As soon as another caller joins, his name will be announced.
Note: Prior to creating the conference, you should notify all of the participants you require to be present in the conference. This notification should include the conference extension number, conference ID, conference PIN, DID and calendar information in the form of an email or any media you deem necessary.
- What is a Hosted PBX?
- Hosted PBX Vs On-Premise
- Hosted PBX Provider – What to look for
- A Hosted PBX for your business?
What is a Hosted PBX?A Hosted PBX is a private branch exchange (PBX) delivered as a hosted service. It’s also known as a Virtual PBX and the advantages companies that use it experience are the elimination of installation, operation and maintenance costs of their VoIP PBX as their VoIP service provider hosts their PBX for them.
A Hosted PBX still allows companies to utilise and take complete advantage of their phone system’s feature set such as voicemail, faxing, automated greetings, conferences and so on.
Companies that offer Hosted PBX’s as a service to their customers handle the call routing or switching as well as all maintenance involved in the hosted PBX service.
There are three ways that a hosted PBX works; either over the Public Switched Telephone Network (PSTN) over the Internet (hosted IP PBX via Internet Telephony or Voice Over IP, VoIP) or with a combination of the two.
Key benefits of a Hosted PBX Solution
- Initial and Ongoing cost savings – there’s no need for a large investment before installing your phone system and also the maintenance cost of your phone system is much lower as you don’t need to hire someone inhouse to operate and maintain your PBX.
- Eliminate office boundaries – employees can work from anywhere with a virtual office phone system. Ideal for home-workers and salespeople who are constantly on the move.
- Boost of corporate image – by using a hosted phone system small businesses can still give the impression of being a large organization with a professional phone system.
- Scalability – with a virtual phone system businesses can add or remove lines and extensions as they grow
- Easy to use – since there’s no end-user management involved with a virtual PBX, it’s easier to use than a traditional phone system.
Hosted PBX Vs On-Premise PBXMany companies are in the dilemma whether they should purchase an On-Premise PBX or a Hosted PBX, the battle of the Hosted PBX Vs the On-Premise PBX begins. It all depends on the size of the company and what they wish to gain from their PBX.
To make a decision, a clear picture of both PBXs should be available. Below are the advantages and disadvantages of a Hosted PBX and an On-Premise PBX.
- A third party provider handles the PBX and all responsibility of running and upgrading the Hosted PBX is shifted onto them
- There are less installation, operation and maintenance costs involved with a Hosted PBX
- There’s no need to worry about any network issues, like bandwidth, with a Hosted PBX; especially important for SMB’s which don’t have the bandwidth to accommodate Unified Communications and VoIP
- Dedicated personnel to manage the phone system are not required; less costs
- No training required on how to run the phone system, how to add extensions and so on. The PBX provider does all that for the end user.
- All upgrades are included in the maintenance costs and are automatically done by the PBX provider
- The end-user has more control over their phone system
- Provided with the ability to integrate the on-premise PBX with other software systems the company is running. For example, CRM systems.
- Very ideal for large organisations which already have the infrastructure, bandwidth and network setting required to host their own phone system, unified communications platform and VoIP
- There’s no need to rely on the support of a third party PBX provider for simple tasks like adding extensions
- All the phone system data and settings are in the end-user’s hands
What to look for in a Hosted PBX Provider?With a Hosted PBX companies essentially take their phone system off-premises to a third-party PBX provider and let them handle much of the responsibility.
The biggest benefits are the cost and the network issues that arise when a small company wants to install a Unified Communications solution. Small companies don’t have the bandwidth to put VoIP and UC on their internal networks so all of a sudden they have to worry about quality of service. Hosted PBX Providers eliminate this headache for businesses. The Hosted PBX Provider still has to be managed by the business but they will not need to hire a dedicated person to do so.
For small companies with 5 to 10 connections hosted PBXs are ideal as they do not have any in-house administration to worry about and neither are they involved in the phone system. The responsibility and administration is passed on to the Hosted PBX provider.
As with any service provider, it’s up to the customer to do their research and compare hosted VoIP/PBX providers to find the one that best fits their needs.Initially companies must look at finding a hosted PBX provider that supports mobile initiatives. It’s important for the desktop phone system to be able to interact with smartphones or other mobile devices to forward calls, but there are also in-depth applications available from some vendors that will essentially give the user a full desktop client on their mobile device. This is particularly helpful for companies that embrace bring-your-own-device policies and want to give users the ability to use mobile VoIP/PBX clients.
It’s also important to find a provider that the company feels that they can trust and that can provide them with personalized customer service.
For example, the company may want to have different queues that are dealt with in different ways, or they may want different forwarding rules. If the hosted PBX provider can’t deal with those requests because they’re too complex, then it’s not going to work. Find a provider that has been around for a while with a good footprint in the market but is still small enough to offer that service.
Why You Should Choose a Hosted PBX for Your BusinessWhen choosing a phone system for your business, the toss up between an on-premise solution or a hosted PBX can be somewhat confusing. A hosted PBX (private branch exchange) is a virtual phone system hosted for you by a service provider.
With a hosted PBX, introducing a whole new phone system to your company isn’t as scary as it sounds. It’s quick and easy to set up so there’s no down time for your business or expensive installation costs. Moreover, you can be safe in the knowledge that you’ll receive the best service and support and that your business’ communications are in safe hands.
More and more companies are opting hosted phone solutions and with good reason, so what are some of the benefits of hosted business VoIP?
- Lower setup costs – as the PBX is virtual, the initial investment is minimal as there is no need for the costly hardware and installation that a traditional PBX requires.
- Elimination of maintenance costs – your PBX is hosted by us so there is no need to hire someone in house to manage and maintain it.
- A scalable solution – hosted PBXs are easy to scale up or down with minimal work and costs.
- User-friendly – For end-users, a virtual PBX is conveniently simple to use and manage. There’s no need for any training or specialised skills so you can start using your new phone system right away.
- Unbeatable mobility & global potential – with a hosted PBX, employees can work from anywhere. All they need is internet access. Additionally, multiple offices can be connected by the same system eliminating inter-office call charges.
What is T38 Fax?
T38 is a protocol that describes how to send a fax over a computer data network. It is needed because fax data can not be sent over a computer data network in the same way as voice communications. Read our How does FAX work in VoIP environments? article for more information. In essence, with T38 a fax is converted to an image, sent to the other T38 fax device and then converted back to an analog fax signal. Most VoIP Gateways and ATA’s now support T38 reliably.
T38 is described in RFC 3362, and defines how a device should communicate the fax data. In the diagram above both the gateway and the fax machine behind the gateway would have to be T38 capable. For the G3 fax machine on an analog line, this process will be transparent. The analog fax machine does not need to know T38.
3CX includes a full featured T38 fax server that allows faxes to be received from anywhere in the network. Faxes can be received as PDF and forwarded via email.
What is FoIP – Fax over IP?
FoIP or Fax over IP (also known as virtual fax) refers to the process of sending and receiving faxes via a VoIP network. It automatically converts incoming faxes into PDF files and sends them to your email account. It is also possible to send an email directly to your virtual fax machine. FoIP allows the smooth transition between technologies, as old numbers and communication methods can be kept while eliminating unnecessary overhead.
How Does FoIP Work?Fax over IP works via T38 and requires a T38 capable VoIP Gateway as well as a T38 capable fax machine, fax card or fax software. Fax Server software that can talk ‘T38’ allows the great Unified Communications feature, Fax to Email, which sends faxes directly via a VoIP gateway and converts the fax message into an email. The plus side is that no additional fax hardware is needed for the Fax to Email feature to work seamlessly!
3CX includes a full featured T38 fax server that allows faxes to be received from anywhere in the network. Faxes can be received as PDF and forwarded via email. Other fax servers currently in the market require the use of separately licensed and expensive Dialogic SoftIP drivers.
How Does FAX Work in VoIP Environments?FAX was designed for analog networks, and can not travel over a digital VoIP network. The reason for this is that FAX communication uses the analog signal in a different way to regular voice communication. When VoIP technologies digitize and compress analog voice communication it is optimized for voice and not FAX signaling. Subsequently, there are a number of things you need to take note of when you move to a VoIP Phone System.
If you want to continue using your old fax machine, and you want to connect it to your VoIP phone system, its best to use a VoIP Gateway and an ATA (Analog Telephony Adapter) that supports T38. T38 is a protocol designed to allow fax to “travel” over a VoIP network.
It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. 3CX Phone System for Windows includes a full featured fax server that is able to receive faxes and forward them in PDF format to e-mail.
Another way to deal with fax when you switch to a VoIP phone system is to connect the fax machine directly to the existing analog phone line and bypass your VoIP system.
Fax to Email
As FAX wasn’t designed for digital networks, FAX to email can be tricky unless you’re using 3CX Phone System for Windows. 3CX uses FoIP – Fax over IP which allows it provide the fax to email service. As part of 3CX’s Unified Communications features, which also includes Voicemail to Email, Fax to Email is widely used by many businesses worldwide as it’s a fantastic way to receive faxes when you’re on the go. The advantage of using Fax to Email with 3CX Phone System is that you don’t need to install a fax modem, fax server or any additional phone lines. 3CX converts the received fax and simply forwards it to your email.
Easily Configure 3CX Phone System for Fax to EmailAs long as your VoIP Provider uses T38, or you have a T38 capable VoIP Gateway, you can receive fax to email easily with 3CX Phone System. Receiving faxes via email is only available in the Standard or Pro edition of 3CX Phone System.
What is RTP – Real-time Transport Protocol?TRTP – short for Real-time Transport Protocol defines a standard packet format for delivering audio and video over the Internet. It is defined in RFC 1889. It was developed by the Audio Video Transport Working group and was first published in 1996. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. RTP is one of the foundations of VoIP and it is used in conjunction with SIP which assists in setting up the connections across the network.
What is SDP – Session Description Protocol?SDP is the abbreviation for for Session Description Protocol. The Session Description Protocol defines a standard for defining the parameters for the exchange of media (often streaming media) between two (typically) endpoints. It has been published by the IETF as RFC 4566.The SDP is typically embedded or encapsulated within another protocol, with the most widely-used application being inside of the SIP protocol inside most IP Telephony applications. In simple terms, the SDP protocol is a declaration, by a media endpoint, of its receiving specifications and capabilities; a typical declaration would tell us:
- which IP Address is prepared to receive the incoming media stream
- which port number is listening for the incoming media stream
- what media type the endpoint is expecting to receive (typically audio)
- which protocol the endpoint is expecting to exchange information in (typically RTP)
- which compression encoding the endpoint is capable of decoding (codec)
…and possibly more. In a typical session setup process, we would see two endpoints participating in a session, where each of the endpoints sends an SDP to inform the other endpoint of its specifications and capabilities. SDP does not in itself deliver any media, but simply limits itself to the negotiation of a compatible set of media exchange parameters; the media streams themselves are handled by a different channel and protocol. Looking at this simple example:
o=MyStreamer 2398026505 2307593197 IN IP4 10.20.30.40
s=MyStreamer Audio Session
c=IN IP4 10.11.12.13
m=audio 15010 RTP/AVP 0 101
…we can see that the endpoint is declaring that it wishes to receive media
- on IP Address 10.11.12.13 (the “c=” parameter)
- on port number 15010 (declared in the “m=” parameter)
- of type audio (declared in the “m=” parameter)
- via protocol RTP, with 2 possible codecs numbered 0 and 101 respectively, where:
- codec 0 is defined as PCMU at 8000hz
- codec 101 is defined as “telephone-event” at 8000Hz (telephone-event is effectively a DTMF tone)
- in bi-directional mode (“a=sendrecv”)
What is H.323?H.323 is, much like SIP, a protocol designed for the setup, management, and termination of a media session. It is one of a set of standards from the ITU-T, which defines a large set of protocols to provide audio and visual communication over a computer network.
H.323, like SIP, is a relatively old protocol, but has been largely superseded by SIP (Session Initiation Protocol). One of the advantages of SIP is that it’s much less complex and resembles the HTTP/SMTP protocols. In this respect, H.323 is a binary protocol, making it less technician-friendly in a troubleshooting environment.
H.323 was not designed for easy extensibility, with the result being that new features require more time to be defined, standardized, and implemented. SIP, on the other hand, is designed from the ground up to be extensible, and a SIP entity that receives a message containing features that it does not recognize can simply ignore the features. SIP is therefore much better able to keep up with the current market and technical needs of the IP Telephony world.
We can see how H.323 is falling into disuse by looking at endpoint devices (phones in particular). A few years ago, the previous trend of creating devices that could use both protocols was abandoned by many phone manufacturers. The vast majority of devices available on the market today are SIP-only.
The result is that most VoIP equipment available today, particularly including IP PBXs, only follow the SIP standard.
What is RTCP – Real-time Transport Control Protocol?RTCP stands for Real-time Transport Control Protocol and is defined in RFC 3550. RTCP works hand in hand with RTP. RTP does the delivery of the actual data, whereas RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.
RTP is originated and received on even port numbers, and the associated RTCP communication uses the next higher odd port number. It transports statistics and information such as octet and packet counts, jitter, and round-trip time. An application can use this information to control QoS parameters and choose, for example, to use a different codec.
RTCP does not provide any flow encryption or authentication methods, but such mechanisms can be implemented with the use of the Secure Real-time Transport Protocol (SRTP).
- Can you List all Known SIP Responses?
- What are SIP Phones?
- SIP Call Session Between 2 Phones
- What SIP-based IP PBX’s are Available?
- What is a SIP Server?
- Sip Phones / VoIP Phones Types
- What is SIP?
- What is SIP Forking?
- SIP Methods / Requests and Responses?
- What are SIP Trunks?
- What is a SIP-URI?
Can You List All Known SIP Responses?SIP responses are the codes used by Session Initiation Protocol for communication. We have put together a list of all the SIP responses known.
1xx = Informational SIP Responses
- 100 Trying – Extended search is being perform so a forking proxy must send a 100 Trying response.
- 180 Ringing – The Destination User Agent has received the INVITE message and is alerting the user of call.
- 181 Call Is Being Forwarded – Optional, send by Server to indicate a call is being forwarded.
- 182 Queued – Destination was temporarily unavailable, the server has queued the call until destination is available.
- 183 Session Progress – This response may be used to send extra information for a call which is still being set up.
- 199 Early Dialog Terminated – Send by the User Agent Server to indicate that an early dialog has been terminated.
2xx = Success Responses
- 200 OK – Shows that the request was successful
- 202 accepted – Indicates that the request has been accepted for processing , mainly used for referrals.
- 204 No Notification – Indicates that the request was successful but no response will be received.
3xx = Redirection Responses
- 300 Multiple Choices – The address resolved to one of several options for the user or client to choose between.
- 301 Moved Permanently – The original Request URI is no longer valid, the new address is given in the Contact header.
- 302 Moved Temporarily – The client should try at the address in the Contact field.
- 305 Use Proxy – The Contact field details a proxy that must be used to access the requested destination.
- 380 Alternative Service – The call failed, but alternatives are detailed in the message body.
4xx = Request Failures
- 400 Bad Request – The request could not be understood due to malformed syntax.
- 401 Unauthorized – The request requires user authentication. This response is issued by UASs and registrars.
- 402 Payment Required – (Reserved for future use).
- 403 Forbidden – The server understood the request, but is refusing to fulfill it.
- 404 Not Found – The server has definitive information that the user does not exist at the (User not found).
- 405 Method Not Allowed – The method specified in the Request-Line is understood, but not allowed.
- 406 Not Acceptable – The resource is only capable of generating responses with unacceptable content.
- 407 Proxy Authentication Required – The request requires user authentication.
- 408 Request Timeout – Couldn’t find the user in time.
- 409 Conflict – User already registered (deprecated)
- 410 Gone – The user existed once, but is not available here any more.
- 411 Length Required – The server will not accept the request without a valid content length (deprecated).
- 413 Request Entity Too Large – Request body too large.
- 414 Request URI Too Long – Server refuses to service the request, the Req-URI is longer than the server can interpret.
- 415 Unsupported Media Type – Request body is in a non supported format.
- 416 Unsupported URI Scheme – Request-URI is unknown to the server.
- 417 Uknown Resource-Priority – There was a resource-priority option tag, but no Resource-Priority header.
- 420 Bad Extension – Bad SIP Protocol Extension used, not understood by the server.
- 421 Extension Required – The server needs a specific extension not listed in the Supported header.
- 422 Session Interval Too Small – The request contains a Session-Expires header field with duration below the minimum.
- 423 Interval Too Brief – Expiration time of the resource is too short.
- 424 Bad Location Information – The request’s location content was malformed or otherwise unsatisfactory.
- 428 Use Identity Header – The server policy requires an Identity header, and one has not been provided.
- 429 Provide Referrer Identity – The server did not receive a valid Referred-By token on the request.
- 430 Flow Failed – A specific flow to a user agent has failed, although other flows may succeed.
- 433 Anonymity Disallowed – The request has been rejected because it was anonymous.
- 436 Bad Identity Info – The request has an Identity-Info header and the URI scheme contained cannot be de-referenced.
- 437 Unsupported Certificate – The server was unable to validate a certificate for the domain that signed the request.
- 438 Invalid Identity Header – Server obtained a valid certificate used to sign a request, was unable to verify the signature.
- 439 First Hop Lacks Outbound Support – The first outbound proxy doesn’t support “outbound” feature.
- 470 Consent Needed – The source of the request did not have the permission of the recipient to make such a request.
- 480 Temporarily Unavailable – Callee currently unavailable.
- 481 Call/Transaction Does Not Exist – Server received a request that does not match any dialog or transaction.
- 482 Loop Detected – Server has detected a loop.
- 483 Too Many Hops – Max-Forwards header has reached the value ‘0’.
- 484 Address Incomplete – Request-URI incomplete.
- 485 Ambiguous – Request-URI is ambiguous.
- 486 Busy Here – Callee is busy.
- 487 Request Terminated – Request has terminated by bye or cancel.
- 488 Not Acceptable Here – Some aspects of the session description of the Request-URI are not acceptable.
- 489 Bad Event – The server did not understand an event package specified in an Event header field.
- 491 Request Pending – Server has some pending request from the same dialog.
- 493 Undecipherable – UndecipherableRequest contains an encrypted MIME body, which recipient can not decrypt.
- 494 Security Agreement Required – The server has received a request that requires a negotiated security mechanism.
5xx = Server Errors
- 500 Server Internal Error – The server could not fulfill the request due to some unexpected condition.
- 501 Not Implemented – The SIP request method is not implemented here.
- 502 Bad Gateway – The server, received an invalid response from a downstream server while trying to fulfill a request.
- 503 Service Unavailable – The server is in maintenance or is temporarily overloaded and cannot process the request.
- 504 Server Time-out – The server tried to access another server while trying to process a request, no timely response.
- 505 Version Not Supported – The SIP protocol version in the request is not supported by the server.
- 513 Message Too Large – The request message length is longer than the server can process.
- 580 Precondition Failure – The server is unable or unwilling to meet some constraints specified in the offer.
6xx = Global Failures
- 600 Busy Everywhere – All possible destinations are busy.
- 603 Decline – Destination cannot/doen’t wish to participate in the call, no alternative destinations.
- 604 Does Not Exist Anywhere – The server has authoritative information that the requested user does not exist anywhere.
- 606 Not Acceptable – The user’s agent was contacted successfully but some aspects of the session description were not acceptable.
What are SIP Phones?Simply put, a SIP Phone is a phone that uses the Open Standard “SIP” to set up and manage phone calls. The actual voice is carried over an IP-based network using another Open Standard called “RTP“. Since these protocols are generically termed “VoIP” (voice-over-internet-protocol), these phones are also sometimes called VoIP Phones or VoIP Clients.
SIP Phones can be classified in 2 main categories:
- Hardphones or Deskphones or Hardware SIP Phones
- Softphones or Software SIP Phones
HardPhonesA HardPhone looks like a regular telephone, and indeed behaves as one. However, the hardware is built using network-aware, or more specifically, IP-aware components. The hardphone will connect to an IP-Network using regular ethernet cables or using WiFi. Cordless hardphones are also available, and these devices take another industry standard cordless technology called DECT, so that the phones communicate with a base station using the DECT protocol, while the base station communicates with an IP-PBX using SIP and RTP as their transport protocols.
SoftPhonesA SoftPhone is quite simply what their name implies – a software program that provides telephone functionality. Again, a softphone will, just like a hardphone, use the Open Standards protocols SIP and RTP for call setup and voice delivery. Any computing device such as:
- Desktop computers (Windows, Mac, Linux)
- Tablets (Android, iOS)
- Smartphones (Android, iOS)
…can run softphone programs, providing a very wide array of options from which to choose. Any computer or smart device that has a microphone and speakers (or a headset) can double up as a softphone. The only pre-requisite is an IP-based connection to a VoIP Provider or a SIP Server, often in the form of an IP-PBX.
SoftPhone BenefitsUsing a softphone allows us to make better use of computing resources, but a more important benefit is actually the fact that it is software-based. The limit of the functionality that can be added to a softphone is limited to the software developer’s imagination, allowing him to create powerful visual tools for the user, integrate into other systems using the softphone itself as the intermediary, and so on…
3CX can be used with most popular hardware SIP Phones, but also comes with a completely FREE software-based SIP Phone that showcases the benefits of a softphone, with extended functions which are not possible to achieve with a hardphone.
An Example of SIP Call Session Between Two Phones
A SIP call session between two phones is established as follows:
1. The calling phone sends out an INVITE.
2. The called phone sends an information response 100 – Trying – back.
3. When the called phone starts ringing a response 180 – Ringing – is sent back.
4. When the caller picks up the phone, the called phone sends a response 200 – OK.
5. The calling phone responds with ACK – acknowledgement.
6. Now the actual conversation is transmitted as data via RTP.
7. When the person calling hangs up, a BYE request is sent to the calling phone. 8. The calling phone responds with a 200 – OK.
It’s as simple as that! The SIP protocol is logical and very easy to understand.
What SIP-based IP PBXs are Available?This list shows some of the currently available SIP-based IP PBXs in the telco sector:
- 3CX Phone System – a cross-platform IP PBX that runs on Windows and Linux. Watch the video Why Choose 3CX Phone System. 3CX offers benefits over other vendors, such as:
- Easy to deploy both in-house and in-cloud
- Automation for in-cloud deployment to top-tier hosting platforms such as Google and Amazon
- Best-of-breed phone provisioning automation
- Robust feature-set, including:
- User-based and Group-based rights management
- Security and Anti-Hacking
- Asterisk – a Linux based IP PBX
- sipX – another Linux based IP PBX
- Elastix – Linux Based PBX
- Brekeke PBX, and others.
A number of open-source alternatives are available, but these all suffer from:
- hard-to-use interface
- poor or non-existent provisioning tools
- limited or non-existent commercial support
- feature set NOT driven by market needs, but by the wants of the individuals that contribute the code
What is a SIP Server?
A SIP server is the main component of an IP PBX, and mainly deals with the management of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar.
Although the SIP server can be considered the most important part of a SIP-based IP-PBX phone system, it only handles or manages sessions; more specifically, a SIP Server can:
- Set up a session between two (or more) endpoints (an audio conference would have more than two endpoints)
- Negotiate the media parameters and specifications for the session for each endpoint using the SDP protocol
- Adjust the media parameters and specifications of a session DURING the session (putting a call on hold, for example)
- Substituting one endpoint with a new endpoint (call transfer)
- Terminate a session
The SIP server does not actually transmit or receive any media – this is done by the media server in using the RTP protocol. Within the context of an IP-PBX environment, it is almost always true that the SIP server and its Media server companion reside on the same machine.
Do keep in mind, however, that very-high-volume SIP servers (such as a large VoIP Provider, for example), may separate their Media server to a different machine to better handle the workload, and could also possibly distribute the load to multiple Media servers.
SIP Phones / VoIP Phones TypesA VoIP Phone System requires the use of SIP phones / VoIP phones. SIP phones come in several versions and types:
SIP / VoIP Soft Phones – Software Based SIP PhoneA software-based SIP phone is an application which makes use of your computer’s microphone and speakers or an attached headset to allow you to make or receive calls. An example of a SIP phone is 3CX’s own SIP clients, which are free to use for all 3CX 12 and above users.
3CX SIP clients for Windows, Android and iPhone
Hardware SIP PhoneA hardware-based SIP phone looks and behaves just like a normal phone. However, it is connected directly to the data network, rather than to standard PSTN line(s). These phones have an integrated mini hub, so that they can share the network connection with the computer, which means you don’t need an additional network point for the phone. Examples of hardware SIP phones are snom and Yealink IP phones which work seamlessly with 3CX.
Use an Analog Phone via an ATA AdapterIf you want to use your current phone with the VoIP phone system, you can use an ATA adapter. An ATA adapter allows you to plug in the ethernet network jack into the adapter and then plug the phone into the adapter. Your old phone will appear to the VoIP phone system software as a regular SIP phone. VoIP phones are very inexpensive to buy and can bought online via one of the many VoIP product online shops. 3CX supports all popular VoIP phones as it’s based on the Open SIP Standard. You can even automatically provision most IP phone models too.
What is SIP – Session Initiation Protocol?SIP (Session Initiation Protocol) is a signalling protocol used to establish a “session” between 2 or more participants, modify that session, and eventually terminate that session. It has found its MAJOR use in the world of IP Telephony. The fact that SIP is an open standard has sparked enormous interest in the telephony market, and manufacturers shipping SIP-based phones have seen tremendous growth in this sector.
The SIP Protocol is text-based, and bears significant resemblance to the HTTP protocol. The messages are text-based, and the request-response mechanism makes for easier troubleshooting. The actual data transmission is done by the Transmission Control Protocol (TCP) or the User Datagram Protocol (UDP) on layer 5 of the OSI model. The Session Description Protocol (or SDP) controls which of the protocols is used.
The SIP messages describe the identity of the participants in a call, and how the participants can be reached over an IP network. Encapsulated inside the SIP messages we can sometimes also see an SDP declaration. SDP (Session Description Protocol) will define the type of media channels that will be established for the session – typically this will declare which codecs are available, and how the media engines can reach each other over an IP network.
Once this exchange of setup messages is completed, the media is exchanged using yet another protocol, typically RTP (Real-Time Transmission Protocol). SIP was developed by the IETF and published as RFC 3261, and its flexibility has allowed it to replace almost completely the H.323 protocol in the VoIP world.
What is SIP Forking?SIP forking refers to the process of “forking” a single SIP call to multiple SIP endpoints. This is a very powerful feature of SIP. A single call can ring many endpoints at the same time.
With SIP forking you can have your desk phone ring at the same time as your softphone or a SIP phone on your mobile. For example, you would use SIP forking to ring your desk phone and your Android SIP Phone at the same time, allowing you to take the call from either device easily. No forwarding rules would be necessary as both devices would ring. In the same manner SIP forking can be used in an office and allow the secretary to answer calls to the extension of his/her boss when he is away or unable to take the call.
What are SIP Methods / Requests and Responses?SIP uses Methods / Requests and corresponding Responses to communicate and establish a call session.
SIP Requests:There are fourteen SIP Request methods of which the first six are the most basic request / method types:
- INVITE = Establishes a session.
- ACK =Confirms an INVITE request.
- BYE =Ends a session.
- CANCEL =Cancels establishing of a session.
- REGISTER =Communicates user location (host name, IP).
- OPTIONS =Communicates information about the capabilities of the calling and receiving SIP phones.
- PRACK =Provisional Acknowledgement.
- SUBSCRIBE =Subscribes for Notification from the notifier.
- NOTIFY =Notifies the subscriber of a new event.
- PUBLISH =Publishes an event to the Server.
- INFO =Sends mid session information.
- REFER =Asks the recipient to issue call transfer.
- MESSAGE =Transports Instant Messages.
- UPDATE =Modifies the state of a session.
SIP Responses:SIP Requests are answered with SIP responses, of which there are six classes:
- 1xx =Informational responses, such as 180 (ringing).
- 2xx =Success responses.
- 3xx =Redirection responses.
- 4xx =Request failures.
- 5xx =Server errors.
- 6xx =Global failures.
What are SIP trunks?SIP trunks are telephone line trunks delivered over IP using the SIP protocol. Using this standard protocol, telecom service (VoIP) providers connect one or more channels to the customer’s PBX. Phone numbers and DIDs are linked to the SIP trunk. In many cases numbers can be ported to the SIP trunk.
Benefits of SIP trunkingBut our farewell to the PSTN brings many benefits. SIP trunks deliver:
- Lower monthly Line & DID Rental The monthly fee to have a number of lines installed at your office drops significantly with SIP trunks. And DIDs cost a lot less.
- Lower call charges –There are many SIP trunk providers and competition has driven down call charges significantly. Some SIP trunks even come with unlimited calling.
- Better customer service –Provide better customer service by adding more geographical and international numbers. Quickly and easily add numbers to your SIP trunk and terminate them on your IP PBX – you can give customers more options to dial in at a significantly lower cost. Customers can contact you more easily and sales will increase.
- Move offices and keep the same number –SIP trunks are not bound to a location, so it’s easy to move offices without having to change your stationary or inform your customers. There is no longer any need to pay to forward phone calls to the new offices.
- Eliminate VoIP Gateways –SIP trunks will eliminate the need to buy and manage VoIP Gateways. All phone calls come in via IP. No extra conversion often means better quality too.
- Leverage a modern IP PBX –Modern IP PBX / Unified Communications solutions will give customers increased productivity, mobility and boost sales. Connecting an IP PBX to SIP trunks is much easier than via the PSTN.
- Flexibility –It is easy to add channels to your SIP trunk to cope with increased calls. A simple phone call will allow you to add channels, and often this can be done immediately. Compare that to the delay in having additional lines installed and then having to upgrade your old PBX to handle more lines!
- Correct number of channels –With SIP trunks, you can easily choose the correct number of channels that you need. Using ISDN/T1, you often have to choose to add either 15 or 30 lines. This usually means you end up with expensive extra capacity.
What is a SIP-URI?A SIP-URI is the SIP addressing scheme that communicates who to call via SIP. In other words, a SIP URI is a user’s SIP phone number. The SIP URI resembles an e-mail address and is written in the following format:
SIP-URI = sip:x@y:Port where x=Username and y=host (domain or IP)Note:If you do not specify a port, the default sip port will be assumed (5060). This is shown in the first two examples below. If you have changed the default SIP port to something else then you need to specify it in the SIP-URI (third example below).
The SIP URI scheme has been defined in the RFC 3261 standard. 3CX uses SIP URIs.
- What is an Auto-attendant?
- Benefits of an IP PBX VoIP Phone System?
- How an IP PBX / VoIP Phone System Works
- What is IVR?
- What is DID – Direct Inward Dialing?
- What is a Voicemail System?
- What are the Benefits of an IP PBX?
- What is Unified Communications?
- What is Voicemail to Email?
- What are BLF Function Keys?
- Arrow 3CX’s Unified Communications features
What is an Auto-Attendant?Auto-Attendant (or automated attendant) is a term commonly used in telephony to describe a voice menu system that allows callers to be transferred to an extension without going through a telephone operator or receptionist. The auto-attendant is also known as a digital receptionist.
For a caller to find a user on a phone system, a dial-by-name directory is usually available. This feature lists users by name, allowing the caller to press a key to automatically ring the extension of a user once his/her extension is announced by the auto attendant. If a user is not available, the auto-attendant directs callers to the appropriate voicemail of the user to leave a message. Having an auto-attendant in a phone system is a very useful and cost-effective feature for a business, as it replaces/helps the human operator by automating and simplifying the incoming phone call procedure.
10 Reasons to Switch to an IP PBX
What is an IP PBX?An IP PBX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network.The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness. The IP PBX is also able to connect to traditional PSTN lines via an optional gateway – so upgrading day-to-day business communication to this most advanced voice and data network is a breeze!
Enterprises don’t need to disrupt their current external communication infrastructure and operations. With an IP PBX deployed, an enterprise can even keep its regular telephone numbers. This way, the IP PBX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.
How it works
How an IP PBX integrates into the networkAn IP PBX or IP Telephone System consists of one or more SIP phones, an IP PBX server and optionally a VoIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server. SIP clients, being either soft phones or desk phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider.
Benefit #1: Much easier to install & configure than a proprietary phone system: An IP PBX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as features. Anyone proficient in networking and computers can install and maintain an IP PBX. By contrast a proprietary phone system often requires an installer trained on that particular system!
Benefit #2: Easier to manage because of web/GUI based configuration interface: An IP PBX can be managed via a web-based configuration interface or a GUI, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems have difficult-to-use interfaces which are often designed to be used only by phone technicians.
Benefit #3: Significant cost savings using VoIP providers: With an IP PBX you can easily use a VOIP Provider for long distance and international calls. The monthly savings are significant. If you have branch offices, you can easily connect phone systems between branches and make free phone calls.
Benefit #4: Eliminate phone wiring! An IP Telephone system allows you to connect hardware IP phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly on the PC. You can now eliminate the phone wiring and make adding or moving of extensions much easier. In new offices you can completely eliminate the need for wiring extra ports to be used by the office phone system!
Benefit #5: Eliminate vendor lock in! IP PBXs are based on the open SIP standard. You can mix and match any SIP hardware or software phone with any SIP-based IP PBX, PSTN Gateway or VOIP provider. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.
Benefit #6: Scalable! Proprietary systems are easy to outgrow. Adding more phone lines or extensions often requires expensive hardware modules. In some cases you need an entirely new phone system. Not so with an IP PBX. A standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!
Benefit #7: Better customer service & productivity! With an IP PBX you can deliver better customer service and better productivity. Since the system is now computer-based, you can integrate phone functions with business applications. For example, bring up the customer record of the caller automatically when you receive his/her call, dramatically improving customer service and cutting costs by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.
Benefit #8: Twice the phone system features for half the price! Since an IP PBX is software-based, it is easier for developers to add and improve feature sets. Most VoIP phone systems come with a rich feature set, including auto attendant, voice mail, ring groups, and advanced reporting. Unified Communications features are included, to support presence, video and audio conferences and free calls via the data network. These options are often very expensive in proprietary systems.
Benefit #9: Allow hot desking & roaming! Hot desking, the process of being able to easily move offices/desks based on the task at hand, has become very popular. Unfortunately traditional PBXs require extensions to be re-patched to the new location. With an IP PBX the user simply takes his phone to his new desk – No patching required!
Users can roam too – if an employee has to work from home, he/she can simply fire up their SIP software phone and are able to answer calls to their extension, just as they would in the office. Calls can be diverted anywhere in the world because of the SIP protocol characteristics!
Benefit #10: Better phone usability: SIP phones are easier to use! Employees often struggle using advanced phone features. Setting up a conference, or transferring a call on an old PBX requires detailed instructions. Not so with an IP PBX – all features are easily performed from a user friendly GUI. In addition, users get a better overview of the status of other extensions, of inbound calls, call queues, and presence via the clients. Proprietary systems often require expensive “system” phones to get an idea what is going on on your phone system and even then, status information is cryptic at best.
IP PBX: How an IP PBX / VoIP Phone System WorksA VoIP Phone System / IP PBX system consists of one or more SIP phones / VoIP phones, an IP PBX server and optionally includes a VoIP Gateway. The IP PBX server is similar to a proxy server: SIP clients, being either soft phones or hardware based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider to the desired destination.
At the center we have, the IP PBX. Starting from the bottom, we see the Corporate Network. This is the company’s local network. Through that network, Computers running SIP clients such as the 3CX softphones, and IP Phones connect directly to the PBX. On the left, we see the company’s router/firewall connected to the internet. From there it can connect to remote extensions in the form of computers running the softphones, remote IP Phones, mobile devices running the 3CX Android and iOS clients, and Bridged PBX’s. Using a VoIP provider we can connect to the PSTN network. To the right a VoIP Gateway connects the PBX directly to the PSTN network.
What is IVR / Interactive Voice Response?Interactive Voice Response or IVR is a telephone technology that allows customers to interact with the company’s host system through configurable voice menus, in real time, using DTMF tones.
How does an IVR System operate?In an IVR system, callers are given the choice to select options by pressing digits. The press of the digit on the telephone keypad sends a DTMF tone to the company host system which then selects the appropriate action / response according to the digit pressed.
Where are IVR Systems Used?IVR systems can normally handle and service high volumes of phone calls. With an Interactive Voice Response system, businesses can reduce costs and improve customers’ experience as Interactive Voice Response systems allow callers to get information they need 24 hours a day without the need of costly human agents.
Some IVR applications include telephone banking, flight-scheduling information and televoting.
3CX has a built-in IVR that is designed to boost the competence of any business by increasing flexibility, simplifying processes and reducing costs, at the same time as improving customer satisfaction.
What is DID – Direct Inward Dialing?DID stands for – Direct Inward Dialing (or DDI, Direct Dialling Inward in Europe) is a feature offered by telephone companies for use with their customers’ PBX system, whereby the telephone company (telco) allocates a range of telephone numbers associated with one or more phone lines. DID allows a company to assign a personal number to each employee, without requiring a separate physical phone line, for each, to connect to the PBX. This way, telephony traffic can be split up and managed more easily.
For example, if an organization has 25 employees and each employee has a separate telephone number, or extension, within its physical location, the organization can rent 10 physical trunk lines from the telephone company that will allow 10 phone calls to take place simultaneously. Others would have to wait for an available line and anyone dialling into the system while all 10 lines are in use would get either a busy signal or be directed to a voice mail system. A DID system can be used for fax and voice transmissions.
DID works similarly for VoIP communications. To allow PSTN users to directly reach VoIP users, DID numbers are assigned to a gateway. The gateway connects the PSTN (public switched telephone network) to the VoIP network, routing and translating calls between the two networks for the VoIP user. Calls from the PSTN will be directed to the VoIP user who holds the corresponding DID number.
DID requires that you purchase an ISDN or Digital line and ask the telephone company to assign a range of numbers. You will then need DID capable equipment at your premises which consists of BRI, E1 or T1 cards or Gateways.
What is a Voicemail System?A voicemail system is a centralized system used in businesses for sending, storing and retrieving audio messages, just like an answering machine would do at home. Voicemail systems make a Phone System more flexible and powerful by allowing information and messages to pass between users even when one of them is not present.
How does a Voicemail system work?Each extension in a phone system is normally linked to a voice mailbox, so when the number is called and the line is not answered or is busy, the caller listens to a message previously recorded by the user. This message can give instructions to the caller to leave a voice message, or provide other available options. Options include paging the user or being transferred to another extension or a receptionist. Voicemail systems also provide notifications to users to inform them of new voicemails. Most modern voicemail systems provide multiple ways for user to check their voicemail including access through PC’s, mobile phones, landlines or even through SIP clients running on smartphones.
A voicemail system in a business is essential to keep external and internal communications flowing seamlessly and efficiently. 3CX has integrated a free voice mail system in its IP PBX for Windows. 3CX Phone System for Windows delivers a complete voice mail solution that incorporates Unified Communications by allowing voicemail to be forwarded to the user’s email inbox.
What are the Benefits of an IP PBX?
- Ease of Installation and Configuration
A traditional PBX is composed of proprietary hardware and software management tools. These tools are typically managed over a serial or console cable, and each vendor has different tools for this. An IP-PBX, on the other hand, is a software-based solution. This automatically means that it is much easier to install and configure, because a system administrator is presented a familiar installation and configuration process.
- Ease of Management
Most IP-PBX solutions provide a web-based configuration interface. The obvious benefit to this is that the system administrator has access to the configuration of the system – the configuration tools are no longer hidden away from the system administrator, allowing him to make the changes himself if he so desires.
- IP-Based Means IP Network
Every telephone system needs to have wiring to connect phones to the PBX. But here is the point of an IP-PBX – your office ALREADY has the wiring, because your phones and IP-PBX run on the same wiring that your corporate network is already using. And again, your system administrator already knows how his LAN network is wired into the network cabinet – the phones are simply additional network devices just like any computer on the LAN.
- Receive and Make calls Anywhere, Everywhere
The SIP protocol is an IP-based protocol, and SIP softphones are now available for any smartphone. This transforms your smartphone into an extension on the corporate IP-PBX, so as long as your phone has IP connectivity, it can talk to the IP-PBX – from a coffee shop, from a hotel room, from an airport lounge, from a yacht marina. Be connected – anywhere, everywhere!
- Cost Reduction
You can use the services of a VoIP Provider – because a VoIP Provider is a telco that can deliver telephony over the internet. So the perfect marriage between IP-PBX and VoIP Provider can be arranged with a very simple ceremony – and the ceremony is called “number porting”. And the benefits of the marriage are immediately visible in the form of reduced call costs. Why? Because land-line telcos have been overcharging for telephony since the first “Hello”.
- Compliance with SIP Standards Eliminates Vendor Lock-In'
Today’s mainstream SIP-based deskphones improves your return on investment. If you need to change from one IP-PBX to another, your phones are still usable – this is because the phones talk a universal language called SIP.
- Scalability – No Limits
A traditional PBX was essentially a hardware device sitting in some corner of your office. It would have a number of empty “slots” to add hardware capacity to your system. Each “slot” would allow you to add “x” number of extensions or “y” number of lines. Once the “slots” were full, you would have reached the limit, and the search for a new telephone system would start – but NOT before you find the money to replace it! An IP-PBX does not suffer from this limitation, because software does not have a limited number of “slots”. If the computer it runs on has the horsepower, you can scale upwards at will. Most commercial IP-PBXs permit this by the simple mechanism of updating the licence parameters assigned to the system – no need to touch anything on the system.
- Reporting and Monitoring
Again, the power of software-based solutions really shines through on the reporting and monitoring functions. For the IP-PBX vendor, extracting data from the call records is a relatively simple task. If a reporting feature is requested by the vendor’s user base, then a new report can be provided simply by way of a system update. Live monitoring of activity on the system is another great bonus which web-based management brings to us.
What is Unified Communications?Currently, there is a large number of communication channels, and of different types, made available to technology users. To put a (indicative but by no means complete) list together:
- Telephony (fixed-line, mobile, VoIP-based)
- Audio/video conferencing
- Presence (as an example, consider your list of contacts in Skype, and the relevant icons that show individual contacts to be online or away)
- Social media (think Twitter, Facebook, Vines, Whats App, Instagram, and so on…)
Some of these communication channels are of the “store-and-forward” type, in the sense that the information is delivered in one direction, and remains accessible (almost) indefinitely for the remote parts to view it when he has the time; e-mail is the grand-daddy of this communication style. Others, however, are more immediate, and require rapid response (often interrupting other tasks); telephony is the obvious largest contender in this category.
Each of these different communication channels typically requires its own “app” to access the information being exchanged. As the number of channels we need to give attention to increases, the harder it becomes to manage them all efficiently.
So What is Unified Communications?Unified Communications, often abbreviated to simply UC, is a generic hold-all term to describe the market’s efforts to integrate all the “apps” (and therefore the communication channels) to allow the user to have all this information easily accessible, irrespective of when or where he needs access (home, work, in a car, on a train…), and how he needs access (laptop, tablet, smartphone, internet cafe…).
UC effectively blurs the demarcation lines between the communication channels. For example, a user can receive a voicemail message and can choose to access it through email or any phone. The sender’s status can be seen through presence information, and if online a response can be sent immediately through chat message or video call.
The objective of Unified Communications is to unify and streamline those business procedures that involve human communications.
Voicemail to Email – Unified Communications on the Go!SIP uses Methods / Requests and corresponding Responses to communicate and establish a call session.
With the voice and FAX delivery feature, your voicemail messages are delivered to multiple email addresses with 3CX Phone System for Windows. As part of its extensive Unified Communications features, 3CX customers leverage the Voicemail to Email feature to ensure they’re always reachable, even when they’re on the go! 3CX Phone System converts voicemails into .wav audio files and sends them directly to you.
BLF Function Keys
Some IP phones have BLF function keys. BLF is an acronym for Busy Lamp Field, which is a light on an IP phone that tells you whether another extension connected to the same PBX is busy or not. Depending on the type of phone you have, BLF’s will remain green, meaning the extension is free to talk. If the BLF starts to flash red, it normally means someone is calling that extension. If the BLF function key is red, it means the owner of that extension is on a call.
BLF’s are also helpful when answering another colleagues phone. For instance, if Bianca isn’t at her desk and someone rings her extension, Andy can pick up the call simply by clicking the flashing BLF. Also, before Helen calls Chris, she’ll be able to see if Chris is on a call or not.
Unified Communications Made Easy1. See Presence of Colleagues
The ability to view the status of other colleagues (“Presence”) is a great time saver avoiding unnecessary call transfers or voice mail tags and makes managing and working with remote employees easier than ever. Need some quiet time to finish a project? Customize your status and prevent any annoying disturbances.
- Eliminate expensive voice mail tags
- Avoid unnecessary call transfers that irritate customers
- Visible from all 3CX clients: Mac, Windows, iOS & Android
2. Deliver Faxes & Voicemail to Inbox
Inbound faxes are converted to PDF and forwarded to users via e-mail, without requiring any fax server software. Likewise, voicemails are converted to sound files and forwarded via e-mail.
- Forward voicemails to inbox
- Listen to voicemails without calling in
- Faxes are received as PDF files in your email
3. Instant Messaging / Text chat
Allow employees to communicate together via text chat, without the need to rely on third party internet messaging systems. 3CX users can send and receive text messages via the 3CX Windows, Mac, iOS and Android clients from anywhere.
- No need for third party messaging systems
- Send text messages, links and more at no additional cost
- Available on Mac, Windows, iOS & Android
What is Echo Cancellation?Before we talk about echo cancellation, we should briefly define “echo” in a telephony context.
What is Echo?Echo, in a telephony context, is a situation where a person’s voice in the outgoing stream from a handset microphone is reflected back to the same person in the incoming return stream to the handset’s loudspeaker.
Echo is typically created in one of 2 ways:
- voice coming through a phone’s loudspeaker gets picked up by the same phone’s microphone (sometimes called acoustic echo)
- electrical signals travelling over a sending wire get picked up be an adjacent receiving wire (sometimes called line echo); this can happen in a few different ways, but is typically to do with the electrical characteristic of the sending, receiving, and intervening devices
How Does it work?Echo cancellation is a simple concept – a voice processing engine analyses an incoming voice stream, and simultaneously monitors the return voice stream prior to transmission. If the engine detects a copy of the incoming voice stream in the return stream, it performs a “mathematical subtraction” on the return stream to remove the copy before sending it out for transmission.
Echo cancellation is sometimes combined with silence detection (and suppression), bringing in some bandwidth conservation benefits as well. However, silence suppression techniques are only really relevant in very limited bandwidth scenarios; as bandwidth becomes more readily available, silence suppression becomes less relevant.
What Does ENUM Mean?ENUM stands for Telephone Number Mapping. Behind this abbreviation hides a great idea: To be reachable anywhere in the world with the same number. Not only that but also via the best and cheapest route. The general idea behind ENUM is to unify the international Public Switched Telephone Network (PSTN), with internet addressing. ENUM takes a phone number and links it to an internet address (URL or IP address) which is published in the DNS system, that can be used in Internet Communications. The owner of an ENUM number can thus publish where a call should be routed to via a DNS entry. What’s more, different routes can be defined for different types of calls – for example you can define a different route if the caller is a fax machine. ENUM does require the specific support from the phone of the caller.
You register an ENUM number rather like you register a domain. At present many registrars and VoIP providers are providing this as a free service.
ENUM is a new standard, and is not that widespread yet. Though it looks to become another revolution in communications and personal mobility.
What Different Types of CODECS are There?A Codec is a device or software capable of encoding or decoding a digital stream or a signal for transmission over a data network. There are video and audio codecs. Codecs are divided into two categories. Lossless codecs and Lossy codecs. Lossless codecs retain all the information contained in the original stream thus preserving the audio/video quality in a signal, while lossy codecs reduce the quality to achieve compression but also use lower data bandwidth.
The following is a list of Codecs that are in common use today:
- GSM – 13 Kbps (full rate), 20ms frame size.
- iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size.
- ITU G.711 – 64 Kbps, sample-based. Also known as A-law/μ-law.
- ITU G719 – 32/48/64/128 Kbps, 28 ms Frame Size.
- ITU G.722 – 48/56/64 Kbps.
- ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size.
- ITU G.726 – 16/24/32/40 Kbps.
- ITU G.728 – 16 Kbps.
- ITU G.729 – 8 Kbps, 10ms frame size.
- Speex – 2.15 to 44.2 Kbps.
- LPC10 – 2.5 Kbps.
- DoD CELP – 4.8 Kbps.
- VP8 is codec used for the encoding and decoding of high definition video as a either a file or a bit-stream for viewing. The VP8 codec is – in contrast to the H.264 codec – free for use.
- H.264/MPEG-4 Part 10 or AVC (Advanced Video Coding) is currently one of the most commonly used formats for the recording, compression, playback of high definition video. In contrast to VP8, H.264 is not free.
What do the Terms FXS and FXO Mean?FXS and FXO are the name of the ports used by analog phone lines (also known as POTS – Plain Old Telephone Service) or phones. The expression “POTS” was initially intended as a joke but is now used as common expression in the telecommunications industry.
FXS (Foreign Exchange Subscriber) is the port that actually delivers the analog line to the subscriber. In other words it is the “plug in the wall” that delivers a dial tone, battery current and ring voltage. This is the jack or interface to the phone system which FXO devices can be connected to.
FXO (Foreign Exchange Office) is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the “FXO device”. This port establishes the connection to the analog line (FXS).
FXO and FXS are always paired, i.e similar to a male / female plug. If no phone system is used the telephone (FXO) is directly connected to the FXS port. The port is provided by the telephone company.
What is a TDM PBX?EA TDM (Time Division Multiplexers) PBX is one of the most common types of voice infrastructures as it has been around the longest. A TDM PBX consists of proprietary, self-contained systems as it was designed before contemporary server technology was invented.
Involving a cabinet with numerous different boards that can perform certain functions, for example intercom functionality boards or analog extension boards, the TDM PBX is coming to the end of its life cycle. The TDM PBX boards are only compatible with systems from the same vendor as an overall architecture, locking in its users to use the same vendor for everything.
A TDM PBX requires dedicated staff to be able to manage it as well as extensive maintenance. It is mostly used by companies which have yet to update their network cabling.
The main difference between a TDM PBX and an IP PBX is that an IP PBX uses Internet Protocol to route calls whereas a TDM PBX uses physical switches. Additionally, an IP PBX is scalable, offers no vendor lock-in and can reduce telco costs drastically.
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