Asterisk encryption

July 15, 2015, 7:41 pm

Asterisk encryption

Page Contents

As of now (Jul 2008) Asterisk does not come with released support for voice encryption. Encryption of SIP signalling is supported as of 1.6 and there is be basic encryption support for IAX, but this is hardly documented and has not been put under scrutiny by security experts. (Regrettably, a message that raised some issues about the security of the session key derivation method has not yet received any reply). Therefore the typical method for media path encryption is to use a VPN. Note that SSH tunneling is not a viable method for VoIP media path encryption.

The BSI (German national office for IT security) clearly recommends to a) separate voice and data IP networks and b) has a preference for TLS and SRTP over IPsec or use of end-to-end encryption protocol like ZRTP. Covert use of built-in microphones of hard- or softphones presents one of the many dangers.

Question: With the missing TLS support in Asterisk could we work around by using OpenSER with TLS in front of Asterisk, and then let Asterisk handle SRTP? Will that influence SIP clients behind NAT that need either the SER NAT helper or nat=yes in Asterisk?

Notice: Please note that SRTP, even when deployed with SIP/TLS support, does not provide end-to-end encryption. The PBX is a trusted third party and can act as man-in-the-middle to intercept traffic. Currently only ZRTP-enabled technology provide end-to-end encryption.

Asterisk channel configuration

Asterisk bounty sip encryption
SRTP for Asterisk: see bug/patch 5413 - requires either TLS for secure SRTP key exchange (sDescrptions defined in RFC4568), or use the included MiKey patch instead
- Asterisk 1.8 comes with SRTP and TLS
- HowTo: Setting up Asterisk with SRTP with libSRTP
- See also this thread on asterisk-devel
- July 2007: Brett Bryant has made progress with TLS support
- Apparently the Google "Summer of Code" 2005 produced a first go at a TLS solution for Asterisk, see also

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New Software Releases

July 15, 2015, 10:20 pm

≫ Next: SIP SS7 gateways

≪ Previous: Asterisk encryption

This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

July 2015

2015-07-15 - Thirdlane releases Business, Multi Tenant, and Elastic Cloud PBX platforms V 7.3.3
2015-07-11 - Optima Dialer runs call at only 6 kbps!(Just launched)
2015-07-09 - miniSIPServer V22 is released to support Customized DNS feature.
2015-07-08 - 3CX Releases 3CX Phone System v14 Beta

June 2015

2015-06-23 -

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SIP SS7 gateways

July 15, 2015, 11:09 pm

≫ Next: Voys Telecom

≪ Previous: New Software Releases

SIP <-> SS7 Gateways

SS7 (or C7) is the main signalling for PSTN interconnections. How do we interface between a SS7 network and a SIP network?

Introduction to SS7 Signaling - This free tutorial provides an overview of Signaling System No. 7 network architecture and protocols.
SIP to SS7 Gateway from Terratel This solution is used in different options of conjugation of VoIP and TDM/PSTN networks. Effective integration TDM to NGN and IMS networks.
SIP Network to SS7 Interconnection This scalable solution provides interworking between SIP and SS7 at one or multiple points of presence.
SmartNode 10k SIP-SS7 Gateways from Patton scale from 120 to 32,000 calls. Offers T1/E1, DS3, and STM-1 interfaces and connects your SS7 edge, core and peer zones with emerging H.323 and SIP networks to facilitate your long-term migration plans.
SS7 to SIP Solutions Scalable SS7 to SIP Signaling Gateway Solutions from Teleprime Advanced Communications Division
Janus Systems High-density SS7/ISDN/CAS/SIP Signaling Converter
Cosmos SS7-SIP Gateway Turn-key, high performance SS7/C7 solution from Cosmact. Scales from 1E1 to 64E1 per chassis. Supports single point code across multiple chassis. Flexible built in routing/digit manipulation rules for SS7/SIP conversion.
Verisign offers a SIP to SS7 interconnection service.
SS7/SIGTRAN SIP access service available at: The Voice Peering Fabric.
Dialogic has an integrated media and signalling VoIP gateway that supports SS7, PRI and CAS for PSTN with SIP and H.323 for IP. IMG1010. Dialogic BorderNet 500 Gateways Provides flexibility in connecting to a wide variety of services and equipment, including SIP trunks, PSTN trunks, and legacy, hybrid, and IP PBXs.
SIP SS7 Gateway - Squire Technologies' media gateway provides full call control routing and supports SIP, H.323, MGCP, H.248 and SS7, PRI ISDN, CAS, R2 + see more SS7 VoIP Case Studies
Simmortel SIP Gateway: SS7 Affordable Gateway Solution built on Sangoma Cards. Scalable and Redundant SS7 architecture. 480+ concurrent calls. ...

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Voys Telecom

July 16, 2015, 5:48 am

≫ Next: Voys Telecom SA

≪ Previous: SIP SS7 gateways

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Voys Telecom SA

July 16, 2015, 7:06 am

≫ Next: SIP Speaker——iSpeaker C20

≪ Previous: Voys Telecom

VoIP telephony for businesses

At Voys, we enjoy liberating businesses from complex, outdated telephone systems and the providers that go with them. Freedom is calling!

Voys is a business telephone service provider. Providing Hosted PBX's, SIP Trunks, Phone numbers (International & Virtual). We believe telecoms can and should be different and offer businesses maximum flexibility and control over their Voice over IP (VoIP) telephony solution. Only pay for what you need and cancel anytime you want. No minimum contract period, no strings attached.

For more information visit our website http://www.voys.co.za

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SIP Speaker——iSpeaker C20

July 16, 2015, 7:03 pm

≫ Next: CooVox Series IP Phone System

≪ Previous: Voys Telecom SA

iSpeaker C20

The SIP based audio system iSpeaker C20 utilizes the built-in intercom and paging capability already inherent in most modern IP PBX systems and enhances this to improve end user experience by providing a dedicated high performance digital amplifier on which to broadcast announcements or play background music.

Basic Features
Support G.711,G.722,G.723,G.726,G.729 audio codecs;
Support SIP2.0 (3621)and related RFC protocol;
Support Power over Ethernet(PoE)
The DSP integrated echo cancellation and noise suppression. If special or higher requirement for voice quality is required then ZYCOO hardware echo cancellation module is an excellent choice.
Easy management and configuration from Web interface.
Support encrypted communication;
Online update of software.

Contact us:

ZYCOO China
Web: www.zycoo.com
Tel: +86 (28) 85337096
Address: 7F, B7, Tianfu Software Park, Chengdu, China.

ZYCOO UAE
Web: www.zycoo.ae
Tel: +971 (4) 3798839
Address: #305, Al Riffa Plaza Bldg., Khalid Bin Alwaleed RD, DUBAI, UAE

ZYCOO UK
LS21, Armstrong House, First Avenue, Robin Hood Airport, Doncaster, UK (DN93GA)

ALL THE RIGHTS ARE RESERVED BY ZYCOO Co.,Ltd

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CooVox Series IP Phone System

July 16, 2015, 7:04 pm

≫ Next: CooBill Billing System

≪ Previous: SIP Speaker——iSpeaker C20

Highlights:

No License Fee
Advanced IP PBX features
Automatic Call Distribution(ACD)
Call Detail Records(CDR)
Fax to Email, Email to Fax
Voicemail to Email
Industry SIP Trunk and Digital Trunks Supported
Firewall Intrusion Detection
HD Voice(G.722.) Supported

Applicable Trunks: FXO/ FXS/ GSM/ BRI/ E1/T1

Software: Developed based on OpenSource Asterisk 1.8.x

Hardware: Dualcore CPU; Modular Design (flexbile to change the module)

CooVox Series IP PBXs come in four sizes: U20 / U50 / U60 / U100

CooVox-U20: Support 30 Ext. Users, Max. 10 Concurrent Calls
SDRAM 128MB DDR2; Memory 4GB SD Card

CooVox-U50: Support 100 Ext. Users, Max. 15 Concurrent Calls
SDRAM 256MB DDR2; Memory 4GB SD Card

CooVox-U60: Support 200 Ext. Users, Max. 80 Concurrent Calls
SDRAM 1GB DDR3; Memory 32GB SSD

CooVox-U100: Support 500 Ext. Users, Max. 80 Concurrent Calls
SDRAM 2GB DDR3; Memory 500GB HD

Contact us:

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CooBill Billing System

July 16, 2015, 7:18 pm

≫ Next: Paging Management System

≪ Previous: CooVox Series IP Phone System

CooBill Billing System

CooBill is a billing system developed by ZYCOO Europe and is designed to integrate with our CooVox Series IP Phone Systems.
Its primary purpose is to aid enterprises in managing their telecommunication billing process, and allow them to access a detailed account list of daily calls or to produce bills for customers. Once integration with CooVox IP Phone Systems is complete, CooBill can be used to check billing for each extension, recharge for a selected extension, etc. Prepay and Postpay are available for flexible pay type options.

Contact us:

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Paging Management System

July 16, 2015, 7:22 pm

≫ Next: Cloud Telephony Management System (CTMS)

≪ Previous: CooBill Billing System

Paging Management System is an innovative software which is primarily used to distribute public address via sip protocol over SIP Speakers. According to the simple interconnection with CooVox IP PBX, multiple SIP Speakers will be centrally managed, monitored and configured through paging management system.

Paging Management System allows you to make announcement with specific voice file at the specific time/date. And all the announcement logs are available to check from the paging management system. Moreover, Paging Management System is easily expandable to add more SIP Speakers to the network.

The SIP based audio system iSpeaker utilizes the built-in intercom and paging capability already inherent in most modern IPPBX systems and enhances this to improve end user experience by providing a dedicated high performance digital amplifier on which to distribute announcements or play background music. There are two models now: iSpeaker B20 and iSpeaker C20.

Key Features
Volume Control
Group announcement
Real-time announcement
Public Address for specified area
Public Address in specified time
Remote settings /management/ upgrade
Centralized management of audio files
Centralized configuration of SIP speakers
Real-time and remote monitor speakers status
Friendly Web interface and easy to use
Paging to up to 500 SIP Speakers
Custom service is available (e.g.: emergency announcement)

Paging Management System Application
Applicable in mall, factory, office, school...

Contact us:

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Cloud Telephony Management System (CTMS)

July 16, 2015, 7:33 pm

≫ Next: Call Center Software

≪ Previous: Paging Management System

What is CTMS?

CTMS：abbreviation of Cloud Telephony Management System, is the perfect solution. CTMS consists of three parts: CTMC+ CTN+ IP Phone.

CTMC： (Cloud Telephony Management Center) is a network node management center which has been independently developed by ZYCOO, and can be utilized by VoIP service providers and enterprises users to manage multiple CooVox CTNs (Cloud Telephony Nodes). CTMC provides a multitude of features for CTN including auto-provision, software/firmware upgrade management, status/performance monitoring, warning log diagnosis etc. CMTC is a powerful solution that delivers the features and functionality required to manage and maintain a highly dispersed telephony environment through the use of a single centralized management system.

CTN： (Cloud Telephony Node) is the node of cloud telephony management system and handling the switch of telephony communication. ZYCOO CooVox series IP PBX can be taken as the node after upgrading. CTN is allowed for branch’s administrator to configure the local network connections.

IP Phone：which support SIP protocol can be used in CTMS; especially the phones support auto-provision with ZYCOO CooVox Series IP PBX; all the phones located in different CTN are allowed to be auto-provisioned via CTMC directly.

Benefits & Features:

Centralized configuration and upgrading of CTNs
Monitor system information, configuration and service status
View and backup of system log, operation log and call log
Manage multi-service and user groups based on template
Manage configuration for individual or multiple devices based on user groups
Flexible upgrading control strategy allowing for convenient software and firmware upgrades
Based on TR069 protocol, allowing nodes to pass through private networks
Adopting B/S managing mode to achieve multi-language GUI, humanized management process, and easy operation
Based on Linux which ensures the device is secure and reliable
Password change supported and license authentication available

Contact us:

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Call Center Software

July 17, 2015, 4:16 am

≫ Next: VoIP Origination

≪ Previous: Cloud Telephony Management System (CTMS)

Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

Types of Call Center Software

ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBX auto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

Contact Center Software

For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

Call Center Software Providers

This is a list of call center software providers and developers. Please keep this list in alphabetical order.

Ameyo Contact Center Software is an all-in-one software based communication solution that manages end-to-end customer journeys and consistently delivers exceptional customer experiences. It is a powerful and highly flexible IP-based Call Center Software platform that lets you have a personalized interaction with every customer across multiple channels, thereby driving customer engagement to a level par excellence. ...

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VoIP Origination

July 17, 2015, 4:50 am

≫ Next: VitalVox

≪ Previous: Call Center Software

Please add information to this page about VoIP Origination.

What is VoIP Call Origination?

One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

What is Call Origination?

VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

Required Hardware

The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

How Call Origination Works

Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

Types of VoIP services

There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

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VitalVox

July 17, 2015, 6:05 am

≫ Next: VitalVox Skills and Queues

≪ Previous: VoIP Origination

Page Contents

VitalVox is a provider of hosted call center and PBX services. By handling much of the infrastructure, it can provide an interface to world-class call center software while reducing the management costs to clients. Through their PBX services, VitalVox allows businesses to scale quickly without having to invest in and manage telephony switching hardware.

Hosted ACD

VitalVox offers a full-featured Hosted ACD allowing enterprise-grade call centers to use the Cloud as their telephony platform. As a complete call center ACD system, VitalVox provides

Skills Based Routing
Queue Prioritization
Agent Hot-Desking
Powerful interactive script builder for agent screens
Drag-and-drop call flow builder for IVR menus

Hosted IVR and Visual Call Flow Builder

The Hosted IVR is a powerful tool offered by Vitalvox. It offers a Graphical User Interface (GUI) allowing you to create dialplans, Interactive Voice Response (IVR) systems, and call flows incorporating both telephony functions and call center ACD processes. With it you can interface with internal and external databases and applications. The Dialplan Builder takes the mystery out of building the and inbound service schedules. With its tree-based structure, the Visual Call Flow Builder presents the call flow to you in an easy to read format, while allowing you to rapidly and efficiently trace the structure of the call.

Dialplans are lists of instructions or steps that the call will follow once it reaches the system handling the call. VitalVox offers fully customizable and versatile dialplan capabilities. This gives you full control over the handling and processing of calls. The VitalVox Visual Dialplan Builder enables the user to unleash the combined power of software and telecommunications to easily control and manage calls.

With the Visual Call Flow Builder, you can use your own audio files and external web services to make the hosted IVR truly your own.

Hosted Outbound Dialer

VitalVox offers an self-pacing predictive dialer that is capable of running multiple concurrent outbound dialing campaigns. It is an integral part of our hosted contact center solution. The dialling modes at your disposal are:

Preview - the agent is able to review the contact before clicking dial
Progressive - the contact is dialed at the same time it is presented to the agent. The agent hears call progress.
Predictive dialing - multiple contacts may be dialed per waiting agent. When one connects, it is presented to one of the waiting agents.

All modes of dialling can use the Script Builder technology that allow for sophisticated agent to customer interactions and powerful data collection with adaptive script branching. Our hosted dialer has a complete set of lead management tools for managing your lists. With live dashboards and historical live view of the dialling results, the Outbound Dialer will provide a detailed stats on your agents, list performance and production targets. ...

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VitalVox Skills and Queues

July 17, 2015, 6:27 am

≫ Next: Virtual PRI

≪ Previous: VitalVox

In a simple agent/queue setup, your calls go into the queue and get handled by the next available agent, no matter who. Your call center deserves better from its ACD.

Being able to automatically log agents into the right queues, keep them out of the wrong queues, and prioritize between them is going to be your key to a top notch customer experience. A sophisticated solution will include agent skills, skill levels, and queue priorities. Using skills-based routing and priorities, you can mix and match to get your best agents doing their best work, and make sure your priority callers are handled as quickly as possible.

Skills-Based Routing

Skills-based routing is simply the ability to direct a call to agents who have the right skills to deal with the call. When you choose a queue for a call to go into, you are specifying that calls require a specific set of skills: those that are assigned to the queue. Only agents having those skills should receive calls from that queue. The call can then be handled correctly.

Agent Skills

Agent skills are simply tags that can be assigned to your agents. If Maria speaks Spanish, you can give her the Spanish skill. If she is also trained in Reservations, you give her the Reservations skill. You also assign the skills required to handle a queue to that queue. When Maria logs in to the system to take inbound calls, she would then be automatically logged in to any queue that has Reservations, Spanish, or both as the skills assigned to that queue. If a queue also requires the skill Widgets, Maria would not be logged in to that queue as she lacks that skill.

Skill Priorities

Not all agents have the same level of skill. Maria may be exceptionally fluent in Spanish, while Harvey is only moderately skilled. You probably want Maria to handle as many Spanish calls as possible, and only have Harvey answer the Spanish calls if Maria is already talking. You can set this by giving Maria a higher skill priority than Harvey. On the other hand, with Spanish being a rarer skill than Reservations, you may give Harvey a higher skill priority in Reservations than Maria so that if a Reservations call comes in that is not Spanish, he can handle it. This helps make sure that Maria will be available for more of the Spanish calls.

Queue Priorities

Some calls may just be higher priority than others. You may want to handle Sales calls immediately, with any Reservations calls taking a back seat. Or you may have an emergency support line that takes priority for any agent with that skill. To handle this need, you can set queue priorities. This indicates that if two calls are waiting and an agent becomes available who can handle both calls, the call from the priority queue will get answered, no matter which call was waiting longer.

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Virtual PRI

July 17, 2015, 11:38 pm

≫ Next: DID

≪ Previous: VitalVox Skills and Queues

Virtual PRI services is an optimum solution for businesses that require quality, high capacity local access numbers without having their own facilities. The Virtual PRI provides immediate turnup and allows expansion into new regions and markets with very minimal cost and investment.

Typical applications for Virtual PRI service are:

Call Centers
CallingCard Applications
Conference Applications
High volume incoming calls businesses

Virtual PRI Provider List (In Alphabetical Order):

DID Number Virtual PRI We Offers DID + 30 incoming channels in 23 countries for 39.99$ a month, for Special offer contact support@buydidnumber.com .

Alcazar Networks - Over 3,100 rate centers. Virtual PRI as low as $46.00 for 23 channels. DID numbers as low as $0.10/each. Stop buying from the middle man.

Virtual Number Virtual PRI Offers DID + 30 incoming channels in 23 countries for 39.99$ a month, SPECIAL OFFERING Also Available 100 Channel DID's in Europe at $ 25 a Month , contact support@buyvirtualnumber.com to get details of Special Offering

CallnFax PRI for business - We offer PRI services in several cities, and the customer support your business needs. We offer hands on service custom tailored to your business needs

DIDLIVE Virtual PRI Offers DIDs + 24 incoming channelsor more in the USA and Canada with over 4000 cities available. Call 1-212-901-0800 or visit www.didlive.com for details.

DIDWW The Source for Wholesale DIDs and Toll Free Virtual Numbers numbers with incoming flat-rate channels in 65 countries.

DomesticNumbers Unlimited channels DID + unlimited incoming channels for €19.95/month covering 70+ countries. Also International Free Phone (Toll Free) and Premium numbers available.

MyVNumber - Virtual Phone Numbers and Virtual PRI Numbers with 30 channels per DID. Amazing Cloud PBX phone system platform is included for each account.

Phone2call Offers Virtual PRI with 30 channels (up 30 simultaneous incoming calls) around 25+ countries. You can forward your Virtual PRI to regular mobiles/landlines (PSTN), VOIP, Google Talk, Linphone, Virtual PBX. Only 2 USD/channel. Know why we keep VOIP leadership!

SendMyCall Virtual PRI Offers DID + 30 incoming channels in 26 countries for only 45$ a month. ...

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DID

July 17, 2015, 11:43 pm

≫ Next: DID Service Providers

≪ Previous: Virtual PRI

Direct Inward Dialing Number (Also known as DID or DDI)

DID (DDI) Background

Most businesses have several incoming telephone numbers used for specific purposes. For example customer service, sales, etc. Some have an individual telephone number for each user in the system. In a home setting on the other hand, each telephone number comes in on a different pair of wires typically. This is not practical in a business environment that has many telephone numbers.

Why was DID actually Created?

So DID ("direct inward dialing") was invented as a way to re-use a limited number of physical phone lines to handle calls to different published numbers. In a business with DID, the phone company uses DID signalling to identify the number they are about to connect to the business's PBX. Historically, this was done by pulsing the last 3 or 4 digits of the number being dialed before connecting the number. The PBX would use these DID digits to switch the call to the right recipient.

In modern PBX's, typically, digital methods (example: PRI) are used to do the same thing, ie. supply the "called party" information. But many business's still have old PBX's which use the analog signalling I mentioned before. The type of telephone lines used for analog DID are different than regular home telephone lines. Usually, battery voltage is supplied by the business PBX instead of the telco. Also, the telco signals a new call by bridging the line instead of by ringing the line. The receiving PBX signals back that it's ready to take the call by momentarily reversing polarity of the voltage on the line (this is called "winking" the line)

Old Fashion Way: (PSTN WORLD)

Direct Inward Dialing is used when your PBX telco connection allows direct dialing to extensions within a PBX, using physical lines (or channels on a PRI) on a shared basis. DID service consists of identifying the "called party" by using DTMF or by digital means, before connecting each call. The service can be sent over an E&M Wink T-1 as DTMF and also as D-Channel information on a PRI.

On a PRI connection, the telco can send only the digits that differ between the group number and the extension (often four digits) or the whole number - it depends on the connection to the telco.

DID (DDI) in the new VOIP World

Let's say you buy a phone line from Vonage or some other phone service provider who offers phone service over broadband. The number that they provide to you, in technical terms is a DID number. This is the number that they have assigned to you to connect you to the old PSTN Networks around the world. Any service provider who wants to offer a phone service over IP address, needs to buy DID numbers from his CLEC or any other large service provider like Level 3 in the United States or go to a consortium (company that will take large blocks from many providers and hand them out one at a time)

If you are using an IP PBX like Asterisk, and you want to connect yourself to PSTN so people can call your office, you can either

1) Buy an Analog or E1/T1 card from Digium, OpenVox, Rhino Equipment Corp or Sangoma

2) Buy DID number from DID service provider

DID Service Providers, convert the analog to digital and provide these DID numbers over the Internet, with SIP or IAX2.

Service providers such as
Wholesale DID Numbers, BuyDIDNumber Fax ,Asterisk Supported BuyVirtualNumber

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DID Service Providers

July 18, 2015, 12:14 am

≫ Next: DIDWW

≪ Previous: DID

A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

SMS enabled DID Providers

DIDWW The Source for Local, National, Mobile DIDs and Toll-Free Virtual Numbers Worldwide. SMS Enabled DIDs in Canada, United Kingdom, Russia, Ukraine and growing. Please download and review our Wholesale DID pricelist.
MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.
SIPMarket is providing DIDs in more than 150 countries. Direct numbers with incoming SMS available. Incoming SMS is free.

Algeria

Incorpus TeleNetworks Incorpus provides DID of 50+ countries. Just visit out website and live chat with us for details. Cheap DIDs available at low costs and discounts for bulk orders. No per minute charge. Only monthly and go on
CarryMyNumber.comAlgeria DID /Virtual Phone Numbers at _wholesale rate@$ 4/month with free PBX with the Free forwarding to SIP, Softswitches ,Trixbox ,Asterisk or VOIP. Phone Numbers from over 70 countries available. Free PBX . Unlimited Channel numbers for call centers /Calling Card Providers__. Largest FootPrint worldwide. No Per Minute charges.
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DIDWW

July 19, 2015, 12:40 am

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ABOUT DIDWW

Our success as a company is directly linked to our values:

We believe that success comes from never being satisfied. We constantly challenge ourselves with new ideas and in solving problems, and we are continually seeking innovative ways of optimizing operations and increasing service quality, all to the benefit of our customers.

We are committed to act with integrity every day, in every way and by every employee in our company. This means fairness and respect in our dealings with each other, our customers and our suppliers, meeting our commitments to our customers and complying with laws and regulations.

OUR TEAM

Employees define DIDWW’s future. We respect the diversity of our workforce and treat others as we would want to be treated. We take and accept responsibility and strive to be role models for others.

Our highly motivated, professional and reliable international team is located in Ireland, the USA, Canada, Israel, Ukraine and Lithuania.

OUR SERVICES

DIDWW was founded in 2004 as a provider of global voice origination services.

Today, our international coverage of local, national, mobile and toll-free virtual numbers is the widest available, and over one thousand telecom companies rely on our services globally.

Our customers include Tier1 carriers, Mobile and ILD operators, ITSPs, calling card companies, conference operators and more.

At DIDWW, we continually strive to exceed customer expectations, anticipating their needs and providing advanced voice solutions coupled with a unique service experience and superior value.

Customers use our virtual numbers to create innovative and differentiated products that attract more business, reduce churn and increase end user satisfaction, while achieving swift profitability.

Implementation of our services is simple, allowing a rapid time to market for our customers' voice products. Easy-to-use management tools are provided, and the DIDWW API enables full automation of Virtual Number business processes.

Our services are delivered over a secure, private intercontinental backbone with full geo-redundant capabilities, automated quality and abuse monitoring, backed by a Network Operations and Customer Support Center operating on a 24/7/365 basis.

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phone.systems™

July 19, 2015, 2:44 am

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A fully-featured, cloud-based virtual phone system that allows you to free yourself from the limitations of traditional telephony and become seamless and globally connected. There is no equipment to purchase, maintain or upgrade, and our unlimited inbound and outbound calling plans eliminate long distance charges and billing from third party phone companies.

phone.systems™ is quickly and easily configured via our unique web interface, with drag-and-drop objects being logically connected together to define the operation of your phone system. No special skills or training are required.

Turn your smartphone into a mobile extension of your phone system. phone.systems™ mobile app offers special features that are usually available only to phones located in your office or at your home.

Explore phone.systems

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Asterisk auto-dial out Chinese

July 19, 2015, 10:24 pm

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Asterisk Call Files

Asterisk call files是個純文字檔，將它移至正確的目錄裡，就能透過asterisk自動撥打。Call files 是個很便利的自動撥打方式，而不用透過其它較複雜的asterisk功能如 AGI, AMI, 和 dialplan, 這些得需要一些技術才能實現。

The Asterisk dial plan extensions.conf responds to someone calling an extension on a channel.若你要從外部應用程式來撥出一通電話，有底下幾種方式可以實現。

asterisk 要啟動自動撥號功能有4種方法

使用.call檔，它是個純文字檔，必須放在正確的目錄裡才行
使用 manager API去啟動撥打。詳見 Asterisk manager dialout
使用指令列 Asterisk CLI originate 這個指令
使用 FollowMe這個指令，從 Asterisk 1.4 版本起，它就有能力實現多方通話(create multiple calls) 但它可能會被濫用於外撥(譯者注:我猜是被用來盜打)

See also additional Digium documents.

New in Asterisk 1.8: A new application Originate has been introduced, that allows asynchronous call origination from the dialplan. ...

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