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astpp2020 · 3 years

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What Makes ASTPP Better Than A2Billing Technically?

ASTPP and A2Billing are two popular names in the VoIP industry often referred to, when someone decided to use an open source VoIP billing solution with the class 4 Softswitch features or vice versa. If you are also considering using an open source VoIP billing software solution with the class 4 Softswitch features, then I will recommend using ASTPP. If you are wondering why then read on. In this article, I will share the top reason to use ASTPP over A2Billing by keeping technical aspects in mind.

1. Technologies

The first and foremost thing is ASTPP is a technically stronger platform than A2Billing. ASTPP is developed using different cutting edge technologies and platforms, including OpenSIPs and FreeSWITCH unlike A2Billing, which is developed using the Asterisk platform. FreeSWITCH makes ASTPP a highly scalable, robust, and reliable platform. You can handle 1000 concurrent calls on 2 servers unlike A2Billing, which can handle only 500 calls on 2 servers. By increasing servers, you can handle thousands of concurrent calls with ease if you use ASTPP.

2. Highly scalable solution

ASTPP is also a highly scalable solution. Whether the number of calls increases or some sort of fault occurs in the hardware, you can still be rest assured that your services will not stop. ASTPP offers excellent load balancing and failover support using OpenSIPs as SIP registrars. You must use ASTPP if you are looking for a reliable class 4 Softswitch with a billing solution. Bonus tips: ASTPP offers several other VoIP solutions unlike A2Billing, which only has a class 4 Softswitch with a VoIP billing system to offer.

3. Technical support

ASTPP has an active community, so in case if you need any technical help and if you are an open source version user, you can ask that easily. On the other hand, if you use the ASTPP enterprise solution, you can avail the technical help from experts. The ASTPP community also shares resources like webinars, video tutorials, documentation, etc. to provide active technical support to the ASTPP users.

4. Installation

Let’s talk about the installation, which is the first technical step to use any VoIP solution similar to other software or apps. A2Billing has some manual steps, which you need to follow to install it. Unlike that, ASTPP has an installation script. At your fingertips, you can install ASTPP. If you are using the ASTPP enterprise solution, then it will be installed by the ASTPP experts for you. A2Billing is not compatible with the latest technologies unlike ASTPP, which can be installed on the latest OS and platforms, including, Debian and CentOS.

Due to multiple reasons, ASTPP is better than A2Billing technically as well as from other aspects as well. If you want to know more about the difference between ASTPP and A2Billing, read this blog on the Difference between ASTPP and A2Billing.

To know the technical comparison between A2Billing and ASTPP, please visit https://www.astppbilling.org/blog/a2billing-vs-astpp/

#class 4 softswitch #astpp enterprise solution

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jennabrileyus · 7 years

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Gventure Technology

Gventure Technology Pvt. Ltd will provide the installation, configuration, and maintenance for the VoIP solutions ranging from small basic business systems to large enterprise based solution. Here at Gventure Technology, a VoIP development Company we will provide you with the solution to all of the problems related to the Voice over Internet Protocol. Gventure Technology will be doing an incredible job of transforming your vision into a real, completely functional website, regardless of the project complexity. Services we provide in VoIP development: Asterisk: An open source framework build for communication applications. A communication server made of a normal computer by using Asterisk. IP PBX systems, VoIP gateways, conference server etc. are powered by Asterisk. FreeSWITCH Development: A scalable open source communication platform. It is designed to route and interconnect popular communications protocols using audio, video, text or any other media. It is capable of handling thousands of simultaneous calls. OpenSips: OpenSIPS (Open SIP Server) is a developed Open Source implementation of a SIP server that runs on Linux platforms and play in the infrastructure of an Internet Telephony Service Provider and it includes application-level functionality. OpenSIPS, as a SIP server, is the main component of any SIP-based VoIP solution. Kamailio: The Kamailio SIP server is the main Open Source software program for building SIP services like a SIP proxy, SIP Presence Server, SIP location server and much more. Kamailio used to handle thousands of call setups per second. Reasons to choose Gventure Technology Gventure Technology has a specialized team who have a thorough knowledge of the Asterisk, FreeSWITCH, OpenSips, Kamailio, IP PBX and much more technology related to VoIP technology. We are having a working experience of 8 years in the VoIP system. We offer you affordable VoIP services to keep your organizations to be updated for the future prospect as well. from Updates & News http://www.voip-info.org/wiki/view/Gventure+Technology

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cloudtelephonysoluion · 7 years

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Now You Can Have the VoIP software development cloud IVR solutions FreeSWITCH development company Of Your Dreams – Cheaper or Faster Than You Ever Imagined

Hello and welcome to my new article about “Now You Can Have the VoIP software development cloud IVR solutions FreeSWITCH development company Of Your Dreams – Cheaper/Faster Than You Ever Imagined.” In this article, I will let you distinguish between these things and recommend you a website where you can get these services cheaper and faster. So, let’s see what is waiting for you.

When you look at VoIP Software Development Cloud IVR solutions FreeSWITCH Development Company online, you will get many results, but all of them are not trustworthy. So, it becomes harder for you to take the right decision and ask yourself which way I should go? I am telling you this is a Company (Gventure Technology) which you can trust and walk further with them. I am going to talk about these topics below

· VoIP software development

· Cloud IVR solutions

· FreeSWITCH development company

VoIP Software Development

Gventure is an entity associated with the advancement and plan of VoIP arrangements in simultaneousness with the requests of the customer venture. They are considered as one among the chief and trusted VoIP solution provider for all the undertaking correspondence needs. They endeavor to put the best foot forward and guarantee to give VoIP business solutions for the customer with the most surprising request of precision. Their relentless attempt toward this path has empowered the productive conveyance to provide demanding outcomes that work to profit the client.

As one among the esteemed VoIP organizations, their specialty lies in the way that we make utilization of open source VoIP platforms, for example,

1. FreeSWITCH,

2. Asterisk install,

3. Asterisk IVR,

4. Opensips, and

5. Kamailio

To address the different VoIP requirements.

For VoIP software development Cloud IVR solutions, Cloud Telephony Solution go to Gventure Technology.

Cloud IVR solutions

Cloud IVR gives you the opportunity to automate and control your most essential client touch point or business forms via telephone without managing complex communication framework. Gventure facilitated IVR solution and advancement devices control a scope of voice applications, which enhance the client encounter.

Gventure has an experience of creating cloud hosting architectures, which are high performance and highly trustworthy. Cloud deployment makes it simple to deliver a customer experience, which can retort quickly to changing client demands while controlling your overall costs, making it easier to retain customers and stay modest in today’s business climate.

Their cloud-based deployment options allow your business to leverage latest solutions, without bearing the burden of implement comprehensive cloud-based interaction management, significant upfront capital or additional IT investments, and workforce optimization technologies including inbound, outbound, and blended voice communications. Enjoy peace of mind with total severance and no single point of failure – backed by their world-class uptime service-level contract.

For VoIP software development Cloud IVR solutions FreeSWITCH development company go to Gventure Technology Pvt. Ltd.

FreeSWITCH Development Company

FreeSWITCH is an extensible exposed source cross-platform telephony platform designed to route and interconnects open communication protocols using text, audio, video or any other form of media. It can be utilized as a soft-client, carrier-class Softswitch or even as PBX. It is the comfort of installation and configuration has made it a very user-friendly PBX solution nowadays. It has a standard design which means that new features can easily integrate into the system as additional modules. Unwanted modules can be disabled at the same time.

FreeSWITCH Softswitch can be installed and work with no trouble in any possible framework stage including Windows, has made it a desirable alternative to the VoIP PBX engineers. In spite of the fact that, FreeSWITCH can be managed through a GUI, the structure of its setup indexes and documents influences the next record to get to the administration more easy to use and straightforward to deal with notwithstanding for the beginner. Setup documents are XML-based. The XML composition is apparent and can be effectively caught on. No XML ability is required.

For VoIP software development Cloud IVR solutions FreeSWITCH Development Company goes to Gventure Technology Pvt. Ltd.

Conclusion

Thank you so much for reading this article. For any VoIP software development Cloud IVR solutions FreeSWITCH development company go to Gventure. They will always be at your service. Don’t put out of your mind to share your opinion about the article in the comment section.

#FreeSWITCH development company #VoIP software development #cloud telephony solution #cloud IVR solutions

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findmyfreelance · 7 years

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opensips & astpp with sipML5 Description: We need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish...

#Freelance Business

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loveperfectcreationcollecto-blog · 7 years

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Kingasterisk technology provides Best Voip Solutions

Asterisk Based Voip Solutions

We are very innovative with opensource voip technology based product, we are heartly invite you to contact us and get more details about our products and marvelous applications.

We are working on voip based opensource plateform since 8 years, we are providing our solutions, services and supports on several applications like asterisk, freepbx, a2billing, dialer, freeswitch, opensips ,kamailio, callweaver, hylafax, elastix, EPABX, IVR, Predictive dialers, Voice Mail, Voice Logger, Video & audio conferencing solutions, web conferencing solutions and lots more.

We are KingAsterisk Technologies where we are developing lots of voip based solutions and applications. Application development, research and issues resolution supports are like our blood in vains, we are keep supporting to our client to achieve their goal in their own decided plateform and models.

IPPBX Solution

IP PBX stands for Internet Protocol Private Branch Exchange. IP PBX is a telephone system that is aimed at delivery of the information including voice, video, etc over the network. IP PBX has become a real breakthrough in the modern technologies as it allows transferring various types of data. IP PBX is especially useful for business enterprises that need to maintain constant contact with customer and affiliates that may be far away. IP PBX is the way to monitor your business throughout the world.

It's the ability to make free calls that makes IP PBX so popular today. International phone calls are becoming much cheaper nowadays but the considerable part of expenses that business companies have goes to cover the cost of international and long distance calls. IP PBX offers a cheap telecommunication service that lets you stay connected with people on the other part of the world. Since IP PBX technologies were introduced hundreds of companies have managed to cut down their expenses and have become more profitable.

VOIP Call recording

Call recording

is a feature present in both client and server side software’s provided by KingAsterisk Technologies. This is a use ful feature especially when there are law obligations or for quality control. On the server each call can be recorded (selectable by route/user) regardless of the codec used (all common codec's are supported). The recording is done using separate low priority background thread which doesn’t affect the call quality in any way. These files are usually stored on a separate hard disk to not affect the I/O speed on the primary disk where the VoIP server is installed

The recorded voices are stored in compressed and encrypted format allowing for easy later playback or export by the administrators from an easy to use interface. To enable voice recording for a user, just open the "users and devices" form from the KingAsterisk Manage, select the user(s), switch the "Functions" tab and tick the "Recording" checkbox. You can listen to recorded conversation anytime later from the "CDR records" form using the by selecting the "Recorded Conversations" radio box.

Click to Call

Click-to-call

is a service which lets users click a button and immediately speak with a customer service representative or interconnect two or more telephone "line". The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to

click-to-call

providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel. Click To Call is the solution for all site owners that like to offer a free phone call to their visitors.

The advantage over other proprietary solutions is that these are true VoIP calls that will work with any phones (voip terminals, softphones, your mobile or landline number) through any telecom service provider. For example you can receive the incoming calls on your VoIP phone or softphone (free calls) and if you are away, then the call can be automatically forwarded to your mobile number.

Sound Box Dialer

offers a call center web based application which can help you to increase your productivity, It will give portabilities to your agents and people who can perform smarter and faster way in your teams.

We have experience in dial applications, so we can keep thinking a lot on this field too. It will give your clients more benefits and more knowledge about your performance so you can get more and more business. This is will give you more effective sound systems where your team can perform better, it will give best and accurate results for your performance.

It will allow to transfer calls, manual way as well as faster way as well. You can also monitor all the stuff on live systems too while agents are in actions. You can use this application in various ways and run in various purposes.

CRM For Call Center

Vtiger CRM is a fully open source CRM application. vtiger CRM is widely trusted by thousands of businesses to effectively manage leads, identify quality sales, track marketing campaigns and monitor inventory. Its features include..

Customer support & service functions, including a customer self-service portal

Marketing automation (lead generation, campaign support, knowledge bases)

Inventory Management / Analysis and reporting

We handle VTiger projects and customize the CRM for various industry-specific needs. Our customer-centric approach makes us proficient in VTiger CRM implementation for various industry verticals. We have expertise in integrating CRM & Telephony.

Voice Broadcast

The KingAsterisk Technologies provides an automated mass dialling solution where hundreds, or thousands of numbers are simultaneously dialled, and an automated connection to an IVR occurs when answered by a person.

For more complex solutions, where the IVR needs to transfer calls to a Live Agent, rather than dialling a fixed number of outbound lines the system can be told how many Live Agents are available and it decides how many numbers to dial based on how many Live Agents are busy. There is a further option for these Agents to log into the predictivedialler.net system, which means that the system knows exactly how many agents are available and whether they are ready to take calls. Using this latter mode there is no need to manually adjust the dialling rate on the Broadcast dialler at all.

Custom IVR Service

Interactive voice response(IVR) is a computerized phone system that enables a person, typically a telephone caller, to make a selection from a voice menu. The selection is made using phone keypad entries or voice responses. This interaction allows the individual to communicate with the phone system and thus the computer system. The phone system plays pre-recorded voice prompts and the person typically presses a number on a telephone keypad to select the option associated with the voice prompt

The KingAsterisk Technologies VoIP server has a built in sophisticated IVR module capable of handling all your business needs including callback and forwarding options, phone to phone calls, answering for SMS initiated actions, announcements, etc. The IVR module is associated with campaigns, which can be set to run different scripts (functions).

Billing Solution

The KingAsterisk Technologies VoIP server built-in billing was designed with carrier grade customers in mind. We offer a complete suite of billing and switching solutions that support the whole range of common VoIP business models. Our VoIP Billing platform will allow you to sell advanced features and its configuration is handled by a simple to use user interface. The softswitch allows service providers to efficiently manage and accurately bill all aspects of their end-users’ VoIP usage.

The KingAsterisk Technologies VoIP billing software is built using 100% multi-threaded C++ code, making it a high performance billing engine that can handle millions of calls. The billing process is running on lower priority threads, never affecting the call quality when the server usage is high. The remote management application offers multiple customizable reports including, accounting, revenue, expenses, call history by user and others.

SMS Broadcasting

SMS Broadcaster is a software which allows you to send/broadcast SMS messages to a list of phone numbers. You can type in the SMS Message and SMS Broadcaster will read in a list of phone number from a file on the root of your SD card and broadcast you’re message to everyone in the file.

This is a lot quicker than do it one by one. Especially, it is very useful when you want to send group SMS to your hundreds or thousands friends or clients. It is an effective tool to improve your efficiency and save your time. You put a file named numbers.txt with your numbers on the root of your SD card using the USB cable, Bluetooth, email or a File Manger then you type in your message and click two buttons and SMS Broadcaster will do everything else.

Video Conferencing

Conducting a conference between two or more participants at different sites by using computer networks to transmit audio and video data. For example, a point-to-point (two-person) video conferencing system works much like a video telephone. Each participant has a video camera, microphone, and speakers mounted on his or her computer. As the two participants speak to one another, their voices are carried over the network and delivered to the other's speakers, and whatever images appear in front of the video camera appear in a window on the other participant's monitor.

The KingAsterisk videoconferencing allows three or more participants to sit in a virtual conference room and communicate as if they were sitting right next to each other. Until the mid 90s, the hardware costs made videoconferencing prohibitively expensive for most organizations, but that situation is changing rapidly. Many analysts believe that videoconferencing will be one of the fastest-growing segments of the computer industry in the latter half of the decade.

Fax Solutions

Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.

Fax For Asterisk provides the functionality to send and receive faxes to / from TDM and IP channels — TDM channels are established across Digium telephony boards and IP channels can use regular G.711 audio encoding or T.38 encapsulation.

Faxes transmitted and received by Fax For Asterisk begin and end as TIFF image files. TIFF files may be readily converted into or from other formats using standard Linux command-line utilities.

Call Center

The KingAsterisk Technologies Callcenter can handle huge amount of inbound and outbound traffic, in a secure, reliable way. The KingAsterisk callcenter combines maximum efficiency with easy to use and intuitive interfaces. Separate campaigns can be setup each of them running separately assigned scripts with graphical user interface for both the operators and the supervisors. By defining quotas, you can restrict your calls to well defined target groups (called clients).

All the call-center related statistics can be viewed in real-time. One of the features of the callcenter is predictive dialing. To restrict the operator wait times, the Calls can be prepared on the server side and dropped to operators when they are waiting for it.

Tele Marketing

We provide you the correct telemarketing services which you need for your business in manner of successful inbound and outbound telemarketing campaign. We can provide you the best applications and services to conduct tele marketing businesses

Our experienced telemarketing program managers design, manage and implement highly customized and flexible customer depended telemarketing campaign that meet your business objectives. Leading organizations and startups rely on us for their inbound telemarketing and outbound telemarketing campaign.

You can create any kind of IVR scripts using a graphical user interface from the remote management client. With more than 30 built-in actions it is very easy to build your custom IVR menus within minutes.

Multi tenants

Multi-tenant management is the ability for a cloud tenant to have omnipotence over the instances, data, and networks in their cloud-hosted solution. In terms of an SP VDI solution this means the vDesktops, the master images, the application distribution mechanism (if applicable), patching, user data, vDesktop networks, access policies, pool size, et cetera. Essentially, the tenant’s management portal needs the ability to perform the primary tasks performed in the VMware View Admin Console or XenDesktop Desktop Delivery Controller console.

In addition, the multi-tenant management solution needs to have the ability to securely provide this level of access to multiple tenants. Unfortunately, this is the first hurdle the major players, Citrix XenDesktop, VMware View, and Microsoft VDI trip over. These solutions have one primary console that’s used to manage the entire environment.

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vindaloo-softtech · 2 years

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Outstanding IT Services by Vindaloo Softtech Outdo Its Competitors: GoodFirms

Vindaloo Softtech Pvt. Ltd. is a foremost IT company that offers exceptional tech solutions. Vindaloo provides VoIP software development, MEAN stack development, database services, digital marketing services, e-commerce development, custom CRM development, web app development, cross-platform, and UI/UX design services. The company was founded in 2016 with headquarters in Ahmedabad, Gujarat, India.

Vindaloo focuses on cutting-edge technologies with an innovative approach to deliver excellent results. The dedicated team is highly experienced in the IT and telecommunication industries. They guarantee robust & reliable solutions for every size of business and a reasonable budget. The company extended its services and served its clients worldwide.

The company was established to serve VoIP app development solutions. In 2016, Vindaloo started to offer its software development services using open source VoIP technologies, FreeSWITCH, OpenSIPs, Kamailio, Fusion PBX, etc. Gradually, they increase their services by adding outstanding web apps, mobile apps, and cross-platform development services. They offer full-service customized development solutions according to their client’s business requirements.

They comprehended the importance of UI/UX and concluded that an attractive design is a must for an app. Further, Vindaloo is determined to provide unmatched UI/UX design in the IT sector. The company possesses expert designers & developers to satisfy their global clients to succeed in it. Today, the company has delivered numerous projects and shares a solid bond with its clients. The team believes in maintaining the relationship with each client that drives massive success. It can also be an opportunity to help them again through Vindaloo’s services & solutions.

The company’s facts & figures:

5+ Years Experience

10K+ Hours Experience

100+ Quality Project Delivered

40+ Experts

GoodFirms is a B2B organization that connects IT service seekers with service provider firms. The platform helps service seekers collaborate with the most appropriate partner through exceptional research on IT firms.

Moreover, the company’s team of experienced researchers and reviewers seeks client satisfaction, market penetration, the overall experience in the market, and quality of deliverables. GoodFirms studied all the registered companies based on the three essential parameters: Quality, Reliability, and Ability.

Similarly, GoodFirms also evaluated the services of Vindaloo Softtech. According to researchers and reviewers, the firm proves to be promising in providing distinctive IT outsourcing solutions.

Vindaloo is a trusted company that offers affordable prices yet zero compromises on quality. The company’s mission is to supply suitable services and develop reliable products for its clients. The team has a tremendous vision for the company to deliver a hassle-free experience to their clients during the product development process.

Vindaloo’s VoIP services include consultancy, troubleshooting, installing, configuring, and developing different open-source VoIP software. The company’s VoIP technology service helps in client service communication. Most startups and medium to large enterprises implement VoIP technology and provide improved ROI by cost control & time. The size is not an issue for VoIP technology to produce excellent customized software development solutions.

Why Choose Vindaloo Softtech?

Top-Grade tech expertise

Budget-friendly

Excellent project management service

Flexibility & Reliability

Absolute confidentiality with a non-disclosure agreement

The company’s core solutions are Hosted PBX, IP PBX, FreeSWITCH solutions, OpenSIPs Development, Kamailio Development, WebRTC Development, IVR, Call Center Software, VoIP Billing, Click to Call, Softswitch, A2Billing, and Mobile VoIP Application, etc.

This expertise has helped Vindaloo Softtech attain a leading position among Ahmedabad’s top IT services companies at GoodFirms.

About the Author

Working as a Content Writer at GoodFirms, Anna Stark bridges the gap between service seekers and service providers. Anna’s dominant role is to figure out company achievements and critical attributes and put them into words. She strongly believes in the charm of words and leveraging new approaches that work, including new concepts that enhance the firm’s identity.

#VoIP app development

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vikingelectonics · 4 years

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Viking E-10-IP Handsfree VoIP Entry Phone

youtube

E-10-IP Features:

Commercial, industrial and residential door security

Vandal Resistant

Flush Mount

SIP compatible (see pg 2 for list of compatible IP-PBX phone systems)

PoE powered (class 1,

Automatic Noise Canceling (ANC) feature for proper operation in noisy environments

Network downloadable firmware

Handsfree operation

Extended temperature range (-15°F to 130°F)

Volume adjustments for microphone and speaker

Flush mountable using the included rough-in box or can be surface mounted using an optional Model # VE-5X5 Surface Mount Box

Optional Model # PB-100 Open Source PC Based Telecom Controller

Optional Model # BLK-4-EWP Blue Strobe Light Kit

Specifications:

Power: PoE class 1 (

Dimensions: See Installation and Specifications

Operating Temperature: -26° C to 54° C (-15°F to 130°F)

Humidity - Standard Products: 5% to 95% non-condensing

Humidity - EWP Products: Up to 100%

Audio Codecs: G711u, G711a, G722

Connections: (1) RJ45 10/100 Base-T, (3) gel-filled butt connectors

Compatibility List:

3COM VCX

3CX

Aastra

Asterisk

Atcom

Avaya IP Office

BlueBox

Brekeke

Cisco Unified Communications Manager (CUCM)

Freeswitch

Grandstream

iptel.org

Kamailio

MetaSwitch

OfficeSIP

OpenSIPS

Panasonic

Samsung Communications Manager (SCM)

ShoreTel

Siemens Communications Server (SCS)

SIP Express Router (SER)

sip.antisip.com

Snom PBX

Sonus

Switchvox

Teksip

Toshiba

VoIP.ms

Vonage

$289.07

source https://www.vikingtelecomsolutions.com/viking-e-10-ip-handsfree-voip-entry-phone.html

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jennabrileyus · 7 years

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CNAM

CNAM is an acronym which stands for Caller ID Name. When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can be used to display the calling party's name alongside the phone number, to help users easily identify a caller. There are numerous CNAM lookup services which allow you to pay a small fee to lookup the CNAM of a specified caller (by phone number).

CNAM Lookup Services List:

http://www.bulkcnam.com/ Cost: Only $0.005 per query for carriers or $0.009 for hobbyists! No catch, guaranteed with easy paypal integration. Sign Up for a FREE Account and we will credit you 30 FREE CNAM queries to try http://www.bulkcnam.com/. http://www.calleridservice.com No monthly fees or account minimums and 20 free queries to test our service when you open an account ( instant setup ). Simple HTTP API or Fast AGI that can be placed in your Asterisk dial plan. Also native support for Switchvox PBX systems. Results are never cached so you get up to the minute real-time results. Retail prices are $.006 per query and bulk pricing is available with a volume commitment of at least 25,000 queries per month. Free support and installation assistance is available. www.callwithus.com offers both CNAM ($0.006) and LRN ($0.0003) look ups. No minimums and monthly charges. Simple HTTP API, easy to integrate to Asterisk dial plan. CID(name) Professional CNAM (Caller name) delivery

EVERY LOOKUP IS LIVE FROM THE SS7 (direct from the carrier owning the number)

NO CACHING... EVER!

NO 3rd party data sources

NO monthly fees

NEVER pay full price for unavailable results

Carrier grade, multi-redundant platform

Simple to integrate HTTP API

99.7% caller id name accuracy

Lightning fast query responses (under 500ms)

Volume pricing as low as $0.002 per query

Try before you buy, 100 free dips with every new account

You choose the output, TEXT/JSON/XML

Track sub-accounts

Easy integration with Freeswitch, Asterisk, OpenSIPS, and other open source voip platforms

Easy access and daily downloads to your account activity

Thousands of happy customers

Get CARRIER GRADE CNAM at http://www.cidname.com

www.cnam.info offers both CNAM and a pseudo-CNAM service at a fraction of the cost. Integration with asterisk is as easy as downloading the AGI and adding a single line to your dial plan. EZCNAM - Highest quality CNAM at the lowest prices with easy implementation and great support . . . Lowest pricing around: $0.003 (3/10 penny) each! Carrier-grade data - highly accurate, never cached. No minimum volume commitments or contracts. Further discounts for over 1M per month. SIP, ENUM and HTTPS queries supported. Integrates with Asterisk, FreeSWITCH, and more. First rate customer service and support. Blazingly fast response times. Onsite server option for increased speed. High Availability / 99.99% uptime SLA. Free Trial Your choice for CNAM is EZ! Call us at (888) 392-6261 http://www.multitel.net/ We have multiple SS7 interconnects and are able to provide you with some of the most accurate and up to date results. Pricing is $0.004 per query for our Free tier and it goes to $0.003USD and $0.0025USD for our Standard and Professional tiers. ... from Updates & News http://www.voip-info.org/wiki/view/CNAM

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