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jobskolkata · 3 months ago

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VoIP Engineer || Vice Process Company || Technical Support VoIP Engineer || Technical Support || NOC Engineer || Kolkata || West Bengal

About Company: Recruitment Company is a rapidly growing Business Process Outsourcing Company, headquartered in London, United Kingdom, and an outsourcing branch in Pune, India. They cover the fields of Business Solutions, Recruitment Solutions, and Designing Solutions as their Core Business. They are specialized and deal in inbound and out bound calls, consultancy, Design and marketing solutions.

Mission of the Company: To be a reliable partner for their clients and they intend to offer new solutions and technology to drive B2B and B2C services with great ease.

Hi we are from Ideal Career Zone: Where Your Skills Meet Your Passions. The ideal career zone is the sweet spot where your skills and passions intersect. It's the place where you can use your talents to make a difference in the world, while also feeling fulfilled and satisfied. Finding your ideal career zone can be a challenge, but it's worth it. When you're in the right zone, you'll be more productive, more motivated, and more likely to succeed.

Now the company need some staffs for the post of VOIP engineer.

Job Description

Profile: Technical Support VoIP Engineer / NOC Engineer

Location: Kolkata, West Bengal

Experience: 3+ Years

Profile: VoIP Support Engineer.

Salary Budget: 5 LPA to 11 LPA

Brief description:

Should have Good exposure to GSM, VOIP, SIP, MSRC, RTP, MSRP, RCS.

Should be adept in VOIP Trunk SIP Configuration, PRI Card Installation, Asterisk/ Dahdi/ FreePBX Installation / IP PBX /IP Telephony / IP phone configuration / EPBX /VOIP Gateway /Asterisk server management.

Troubleshooting of SIP and VOIP Based Call-center telephony Issues.

Experience in Installing, configuring and deployments of Asterisk, Asterisk-based applications like Vicidial, IVR and FreePBX.

Experience in Installing, configuring and deployment of Asterisk, GoAutoDial, Vicidial, IVR and FreePBX.

Sound knowledge of Asterisk Installation, Configuration, Dialplan, AGI, AMI, Call troubleshooting (SIP, ISDN, PSTN)

Configuration, Maintenance and troubleshooting of Asterisk Based Servers, VICIDIAL, FreePBX, Freeswitch.

Good verbal and written communication.

Excellent team player, ability to work in a global team and follow through on deadlines.

Strong technical and analytical skills.

Resolve Client issues through Skype and Remote Screen sharing.

Minimum 2+yearsexperience in Troubleshooting VoIP Required.

Proficiency in the following programming languages: MySQL, PHP, Perl. Preferred skills

Industry Type: Telcom, ISP, BPO.

Key Skills: Asterisk, Vici Dail, Free PBX IVR, VOIP, Wire Shark

Employment Type: Full Time

Note:- Many more Jobs available just search in Google “Ideal Career Zone” Kolkata.

You can find many more job details in various posts in various companies.

You may call us between 9 am to 8 pm

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Or you can visit our office.

Ideal Career Zone

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Thank you for watching our channel Please subscribed and like our videos for more jobs opening. Thank You again.

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vindaloo-softtech · 2 years ago

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Asterisk vs. FreeSWITCH: How Are They Different?

Asterisk and FreeSWITCH are the two most widely used platforms for developing VoIP solutions in the VoIP industry. They are both powerful enough for developers to create any intricate VoIP solutions for teamwork and communication in addition to being open-source technologies.

What is Asterisk? In 1999, Mark Spencer created Asterisk, a piece of software that distributes calls like an expensive PBX. His tiny group created the Asterisk call distribution and handling software for their business, Linux Support Systems, which was eventually renamed Digium.

What is FreeSWITCH? The Asterisk platform has drawbacks, which led to the creation of FreeSWITCH in 2006. Anthony Minessale, a well-known Asterisk developer, made the decision to create a software from scratch to address some of the alleged problems with the Asterisk platform. This was later known popularly as FreeSWITCH.

If you are wondering why to switch from traditional to VoIP calling, here is an article on “PSTN v/s VoIP”

How Does Each Technology Work? The heart of every Asterisk system is the dialplan. It is a scripting language, and the modules are used to give instructions to the Asterisk system through the configuration directory. Developers may implement a variety of capabilities, including call reception on a particular SIP channel, call connection to IVR, and dial plan-based call routing. Asterisk's configuration files are stored as standard text files.

FreeSWITCH adopts a different strategy; the system was created in C, and the core programming was better organized. FreeSWITCH uses processing threads that operate uniformly throughout memory, in contrast to the Asterisk approach, which gives each channel its own thread and memory space.

Basic Functionality On the most fundamental level, both technologies offer the majority of the same functionality. Voicemail, call recording, and IVR menus should be available on every FreeSWITCH Development or Asterisk-based switch on the market. With any design, the steps involved in constructing extensions and gateways are quite similar. Nevertheless, depending on the server's Memory and core performance, different numbers of users may be supported. For communication with other Asterisk systems, Asterisk needs the proprietary IAX protocol, but FreeSWITCH is not constrained in this regard.

MultiTenancy One of FreeSWITCH’s features is its capacity to support several tenants. As a result, many user branches may use a single FreeSWITCH system as unique entities under their own domains or subdomains. For multi-tenancy to function on Asterisk systems, costly proprietary solutions must be constructed on top of it at the time of Asterisk software development.

Clustering Asterisk is primarily made to operate on a single system. Using a single PBX server to install all roles is equivalent to placing all of your eggs in just one basket. Thus, it is advised to divide each system, performing a given function, into a distinct server for large-scale or enterprise- level PBX servers, i.e., distributed architecture. While FreeSWITCH solutions makes it simple for separate systems in the cluster to execute individual duties, accomplishing this with Asterisk is difficult.

IM abilities Systems like Asterisk and FreeSWITCH provide cutting-edge communication features including conference, video calling, and chat. Unfortunately, the majority of Asterisk systems require on an add-on for IM functionality, which businesses must pay extra fees for. With FreeSWITCH, the only requirements are that the XMPP service be enabled and that the end devices be correctly set up for IM.

Device Deployment Capabilities On either FreeSWITCH or Asterisk networks, device deployment varies greatly. Several endpoint management modules for IP phones and softphones are supported by Asterisk, however, access to the provisioning software costs around $100. Compared to Asterisk, FreeSWITCH offers a significantly smaller selection.

Wrapping Up There is no discernible difference between a well-setup system running Asterisk or FreeSWITCH for the end user. In fact, FreeSWITCH provides a wider variety of modularity and is a better option for various VoIP solutions. With more than six years of experience developing commercial communication solutions using either open-source VoIP communication technology, Vindaloo Softtech is an accomplished VoIP development firm.

#FreeSWITCH vs asterisk #asterisk vs FreeSWITCH #Asterisk Development #Asterisk software development #Asterisk Solutions #FreeSWITCH development services #FreeSWITCH solutions #open source VoIP communication technology

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cloudsdial · 4 years ago

Link

#vicidial #asterisk dial plan #vicidial lead fields #cloudDial #api dialplan #asterisk dialplan variable

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esatyabca · 7 years ago

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Manipulating Dialplan Variables : Asterisk

We often require to do string manipulation on a variable. For example, a variable named phonenumber which represents a number we’d like to call, and we want to strip off the first 5 digit before dialing the number. Asterisk provides a special syntax for doing just that, which looks like ${variable[:skip[:length]} The optional skip field tells Asterisk how many digits to strip off the front of the…

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#asterisk #asterisk dialplan #Dialplan variable manipulating #Manipulating Dialplan Variables asterisk

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wsitho · 2 years ago

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Inbound Name Centre Services

Using this information, we're able to develop your perfect purchaser persona profile, and establish and address the questions they would doubtlessly need to be answered during each stage of the customer journey. Our team has broken down the inbound advertising process into three primary areas – consideration, conversion, and retention. With our inbound packages, we're always considering of your customers’ next steps and making inbound services sure that the consumer journey is as smooth as potential. Our staff will carefully monitor your inbound advertising results and use suggestions to continually optimize your marketing strategy and goals. All of our packages embrace entry to an inbound advertising marketing consultant whose aim is to implement your technique and take your business to the subsequent degree.

A key issue to think about when selecting ABM over Inbound is the type of buyer you may be focusing on. Small businesses are more doubtless to answer Inbound Marketing and fill out a ‘contact us kind than giant enterprises, particularly when a large funding is required to purchase your services and merchandise. Your inbound advertising company additionally serves as a touchpoint for your whole inbound advertising inbound services questions and special requests. If you have an concept for a new method to attract consumers nearer to your brand, your inbound advertising agency can help you make that concept a actuality. Monitoring every single call is very important for inbound name centers as a result of sustaining consistency and high quality if calls is essential.

At BluCactus Inbound Marketing Agency, we strategically manage and monitor your business’ social media platforms. Consequently, our staff pays attention to what your prospects say about your organization and how they interact together with your online presence. We hear out in your potential clients’ questions, comments, likes, and dislikes – and attain out to them with related content material. Thus, we help you build model consciousness, model loyalty, and improve the overall profitability of your corporation in the digital age. As an inbound advertising firm dedicated to our clients’ satisfaction, our job is to assist your organization grow. To achieve this, we focus on attracting and changing your leads, helping you scale back the price of buyer acquisition, and shortening your sales cycle.

web optimization and SEM both contain totally different methods of promoting web sites online. Expected to achieve $171.9 billion, in comparability with $145.5 billion in 2021. Most of the time, completely different corporations supply extremely similar SaaS merchandise, leaving it as a lot as the marketers to focus on and market their unique selling propositions in front of the purchasers. DeljooSoft’s vision is to supply the latest innovations in software improvement to companies of all sizes and types. We work to make both current and future tasks straightforward for companies with life-changing merchandise that work for firms.

Thus, every potential client who approaches you might be treated with priority and the time needed for a optimistic first expertise that stands out. Before deciding what’s finest for your brand, we are going to outline the terms outbound and inbound and tips on how to use them to generate outcomes and drive sales. The short answer shall be that gross sales activities lie within the outbound category when classic advertising actions are associated to inbound ways. Never assume that you realize your customers’ needs greater than they do. Rather, try to hearken to what your customer feels or says and act on that data in a means that your buyer finds useful. Some will say stuff you don’t like, however you can't simply run or wish them away.

The Q-Suite supplies one of the most interesting ACD for any call heart software and is particularly designed to run on Asterisk. The ACD comes with a robust Dialplan builder for creating refined IVR . Every call routed to an agent will have detailed call pop-up with DID/DNIS, marketing campaign and Queue information. The ACD is linked to the scripts allowing not only the pop-up of the call and queue detail but in addition the related script together with net functions and CRM.

If you want to see results from your digital efforts, you need to think past your inbound marketing activities. Once you've your up to date inbound marketing strategy, we’ll carefully monitor and monitor the results of your campaign to ensure you’re on the right path to progress and success. We’ll also present a transparent and detailed report that lets you know inbound services how you’re performing so you can rest assured you’re all the time staying on high of your game. Videos are another wonderful inbound advertising channel to leverage. Some Facebook and YouTube users have really risen to fame simply using the stay video features. One profit is that persons are more more doubtless to watch a video than read an article.

#digitalmarketing

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elisionvoipsolutions · 3 years ago

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Top Reasons to Use Asterisk Development Service from AC InfoSoft

There are many companies all across the world that use different communication and collaboration solutions developed on top of different VoIP technologies. One of the most popular and widely used VoIP development technologies is Asterisk. Being a pioneer in the industry of VoIP development, Asterisk is a leading VoIP solution development technology. There are thousands of companies that are using Asterisk-based communication and collaboration solutions. This is also the cause of a plethora of options available when someone wants to get Asterisk development services.

There are hundreds of Asterisk development service providers, including, companies and independent freelancers. This makes the process of finding the best Asterisk development service provider tougher. To simplify this process, I have the recommendation to use the Asterisk development service of AC InfoSoft.

The company has been in the industry for more than 12 years. The company has been offering the best Asterisk development services to its customers since its inception. Not only this, but the company also offers some innovative products that are developed on top of Asterisk. In a nutshell, the company has the required experience and expertise to provide the best Asterisk development service.

The company has a team of experienced Asterisk developers and support engineers that have been working with all different types of VoIP solutions developed using Asterisk technology. Below is the list of key features worked upon by the team of AC InfoSoft:

· Call center solution

· IP PBX solution

· Interactive Voice Response solution (IVRS)

· Voice broadcasting solution

· Fax broadcasting solution

· Missed call solution

· Click to call solution

· And more

The company also offers Asterisk development services at affordable rates. It means anyone can get benefited from expert Asterisk development services of this company. AC InfoSoft has catered to hundreds of customers till now and the company is offering the best in the industry support services after the development of the solution to stay by the side of its customers on the journey of growth and development.

AC InfoSoft also offers the “hire an Asterisk developer” model for the customers that want to hire Asterisk developers for full-time or part-time that work dedicatedly on their project.

Along with development, it also offers Dialplan programming and AGI scripting services. To know more about development and other services offered by AC InfoSoft in Asterisk technology, please visit https://www.acinfosoft.com/asterisk-services-solutions/

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crmvoipsoftwaresolutions · 3 years ago

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AC InfoSoft Announced to Offer Custom Asterisk Development Services

AC InfoSoft announced to offer custom Asterisk development services for its global clients. The company will offer diversified solutions and services in Asterisk.

AC InfoSoft is a renowned name in the VoIP realm because of its innovative solutions and client centric services. The company has announced to offer Asterisk development for its clients all across the globe. The company has a team of experts that have been working on Asterisk for many years and have developed robust, secure and scalable Asterisk solutions. The company thrives to benefit its prospective customers with this expertise.

The company will cater all different sized businesses with its Asterisk software development services. The company has claimed to offer Asterisk development services at lower cost, so anyone with a limited budget can also take advantage of its services.

The representative of the company announced that the company will cater all different types of requirements, including, but not limited to:

Asterisk software development

Asterisk solution customization

Asterisk application development

Custom IVR development in Asterisk

Asterisk module development

Asterisk call center solution development

Asterisk voice broadcasting solution

Dialplan programming

AGI scripting

And more

As per the announcement made by the representative of the company, the team of Asterisk solution development has catered many clients till now. They have required skills to develop any custom solution in Asterisk. Along with the custom development and customization services, the company also offers support services to benefit its customers.

��The Asterisk is the pioneer VoIP development platform. It is furnished with all required features to develop any type of simple to complex VoIP solution. From a simple calling system to a multitenant conferencing solution can be developed on top of Asterisk. However, to make correct use of its power, one need Asterisk expert. We have ingenious Asterisk developers that have made their hands dirty for years in this technology. We would like to benefit our clients with their expertise to make their business grow as we aim to become a technology partner of our clients.” the spokesperson of the company shared.

The company follows a well defined model to cater Asterisk solution development needs of their clients. Also, it uses the latest project management tools to keep their clients updated about ongoing development progress. The company has enthusiastic business developers that are able to align the Asterisk software development projects according to the customer’s preferences and point of view. Furthermore, the company has kept its rates very competitive to benefit all different scaled businesses.

The company has launched a webpage on their official website to showcase its Asterisk development services to their website visitors under the services section.

About AC InfoSoft It is a VoIP company. The company caters all different IT arenas to benefit its clients. VoIP is the core expertise of the company. The company caters to many customers worldwide with its various services in the VoIP arena and Asterisk development is one of the offerings of the company. Visit https://www.acinfosoft.com/asterisk-services-solutions/ for more details.

#Asterisk development services

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robustkite · 8 years ago

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Setting up a SIP intercom system with Cisco 79xx Phones

So a while ago I purchased some old Cisco IP Phones from an office that was clearing out its old stuff. I was interested in setting up some phones around the house as a kind of internal intercom system. After a week of fiddling and finding out whatever info I could about the models I had my hands on (7941, 7942, and 7962), I finally have them calling each other and working with FreeSWITCH.

I also managed to get images appearing on the screen too! Nifty! Other images I put together for other phones can be found here.

Another great thing is it also breathed some life into my old android phone too - it’s now a wireless intercom, using CSipSimple. Combined with an audio loopback on my PC, I can stream my desktop’s audio straight to a conference call using MicroSIP - now I can afk and still be listening to podcasts, talks, etc...

I certainly learned new skills, and a little about how this phone stuff works!

(Feed Your Inner Techician: For more info on some things I learned in setting this all up, read on below)

I certainly had a bunch of pages to look through, though some provided more useful info than others - I can’t remember all of them haha. But I’ve put a list of some useful links for anyone who’s interested.

Note that this is by no means a complete guide! It’s just some useful info I learned, and I’m also writing it as a means to consolidate the stuff I came across. :)

Key things in my set up:

Something that works as a SIP Server (i.e. FreeSWITCH, or Asterisk both open-source).

Software SIP Client(s) (such as CSipSimple (Android), MicroSIP (Windows), Linphone (cross platform, mobile and desktop versions available) - I used MicroSIP for Windows because it’s portable/no installation required)

Hardware SIP Clients (Cisco 7962, 7942, and 7941 IP Phones) - mine didn’t come with power cords as they were Powered over Ethernet and none of my devices supplied PoE, so I got some random, no-name PoE Injectors I searched up on eBay.

(Only for the Hardware SIP Clients) A TFTP Server such as Tftpd32 (or Tftpd64, which is the 64bit version), which will be sending configurations to the Cisco Phones when they boot up.

Ethernet Cables :P

Tftpd32/64 and Cisco Firmware Files

This little standalone program serves out things like the firmware files for the 7941/7961 or the 7942/7962 phones. They also have the firmware for various other models too.

This is also used to serve out the configuration files to the phones - In my googling about configuration files, I learned that some models use a .cnf extension and format (which is older, I think), and others use an XML format.

Fortunately the configurations for the 79x1 and 79x2 both used the XML layout for their configurations, so I didn’t need to pick up 2 config layouts/structures, though I suspect they would follow a similar structure/pattern.

Tftpd and its DHCP setting

I recall a few guides I read on getting Cisco SIP stuff set up saying that you’d need to turn on Tftpd′s DHCP feature, as well as setting something like option 150 and providing an IP address in Hexadecimal...

For some reason I couldn’t get that working (then again, the posts were like.. 2009? to.. something like 2013?), and the phones weren’t getting IP’s set from Tftpd’s DHCP service.. so I just used my router’s DHCP service instead - in its settings it had a field labelled “TFTP Server Name (Option 66)”, once I filled that in, the phones managed to connect and download their firmware (I was using v9.2.1).

Initially these were using Cisco’s “SCCP” protocol, which I think both FreeSwitch and Asterisk both support, but by the time I figured that out I already had put the SIP firmware onto the phones already. :P

So.. Apparently I didn’t need to use Option 150? Not sure why, but it all seemed to work. Just something to keep note of if you want to try and run into a similar problem even if you follow the guides out there which talk about using option 150.

Cisco 7941/7961, 7942/7962 Configuration Files

I found that the Tftpd server’s logs show that the phones require multiple files when they boot. This list doesn’t include the firmware files and therefore assumes the firmware is already loaded on the phone.

(Note: If the phone receives a configuration which defines a different firmware version than what it has previously loaded, it will fetch it from the TFTP server and flash itself anew. Otherwise, I’m pretty sure if it times out for whatever reason, it will continue to boot to the last firmware that was loaded on it)

The files it seems to ask in the Tftpd log are

CTLSEP<Phone’s MAC Address>.tlv

ITLSEP<Phone’s MAC Address>.tlv

SEP<Phone’s MAC Address>.cnf.xml

dialplan.xml

What I’ve found is that CTL relates to a “Certificate Trust List”, according to this article from Cisco. The phone seems to function fine without it (for SIP at least, dunno about SCCP), and I ended up just making a blank file in the TFTP server’s folder to avoid seeing “Error file not found” logs coming up.

As for ITL, I couldn’t quite figure out what that was - maybe another Trust List? Since I couldn’t get info on it in a reasonable amount of time, I ended up just making a blank file for that too - haven’t seen any problems coming up from this either. :P

The last two, (the XML ones) are probably the main things that affect the phone’s functionality in my case, since the SEP<MAC>.xml file contains all the configuration info such as the SIP server’s location (eg: an IP Address), ports, preferred codecs for calls, auto-pickup feature, speed dial buttons, text that shows up on its screen, extension, authentication, and plenty of other things.

SEP<MAC>.cnf.xml

This article at Brain Overload > SEPXXXXXXXXXXX.cnf.xml nicely documents a lot of settings in the XML file. Definitely bookmark or save if you want to tinker with the settings and want a reference.

I just tinkered with settings for a while until I got something that was working in terms of display and whatnot. One thing I couldn’t figure out for days was why they wouldn’t “register” with FreeSWITCH. Phones need to “register” to the SIP Server so that when someone calls an extension, the server knows ‘where’ to forward the call stream to (i.e. IP address, etc.). If a phone doesn’t register, a caller gets sent straight to Voicemail.

So during these troubleshooting days, I was able to do the following:

Make calls from the phone (eg: calling Voicemail, trying to dial an extension, dialing 9198 and hearing Tetris music streamed back from FreeSWITCH, etc... the FreeSWITCH log/window shows it’s all working.

Leave voicemail for an extension

Call voicemail and successfully retreive/play back voice mail for extensions

I just couldn’t receive calls - FreeSWITCH’s logs shows “Originate Failed. Cause: USER_NOT_REGISTERED”, which also coincides with the phones constantly showing the status “Registering”.

Turns out it had to do with the section in bold below (example code from Brain Overload)

...

...

For some context on this section, “<sipLines>“ starts the section that defines the different front buttons to the right of the screen on the 79xx phones - and is required for the phone to function.

All the guides that were top hits in my googling always had an IP Address in the <proxy> tags. Makes sense, of course - Line 1 will talk to the proxy at whatever address is defined there.

Days later I found that another valid entry can be placed in the <proxy> tags:

<proxy>USECALLMANAGER</proxy>

So a lot of examples I saw out there in guides I would see this section, every time (because the phone wont work without it):

<ports>

</ports>

</callManager>

I always wondered what this section did - when I tinkered with it, it didn’t seem to affect much (i.e. it always would stay on the “registering” state). Guess changing the <sipLines> section to “USECALLMANAGER” did the trick. (Hope this might help someone else out there if you run into a similar problem like I did).

DialPlan.xml

The dialplan.xml lets the phone send the entered phone extension/number to the server after a certain time, and after matching a pattern/rule defined in the XML. Search for the title “DialPlan Notes (dialplan.xml)” in this article Voip-info.org > Asterisk phone cisco 79xx for example XML for a dialplan and notes on it too.

For reference the main line I used to test things out were the following entries:

The first two MATCH’s correspond to FreeSWITCH’s default extensions/accounts that it has when it’s freshly installed (i.e. 1000, 1001, 1002... 1009, 1010, 1011... 1019). These are rules whereby once the user enters either “100″ or “101″, and any digit (i.e. the dot represents any number from 0 to 9), the rule will be matched and the phone will immediately (i.e. timeout of zero) send the call/extension request to the server to ring that extension.

The last entry has four dots in the MATCH tag, and 2 seconds time out. This indicates any 4 digits will match this rule after 2 seconds of no other numbers being dialed. So if you enter a number like “2000″ for example, the phone will not send the data/request to the server until after 2 seconds.

Keep in mind you can remap extensions too, sort of like a speed dial:

This will effectively translate dialing “19″ to call the extension of “1019″ instead. The 4 examples will also work if they are all present in the dialplan too, just keep in mind that if you have “10″ as a match (with a timeout of zero), it will always trigger before you ever get to “100.”.

NOTE: Keep in mind these rules only trigger when the line is “active”, i.e. you pick up the handset or switch on speaker. Entering numbers before selecting a line wont trigger the rules and dial numbers.

FreeSWITCH

I had initially tried Asterisk, my first impression of it was a bit intimidating, and I didn’t quite know where to start - perhaps someone who is more familiar with PBX/SIP stuff might have, but the thing I wanted to achieve was just an internal phone setup... So I tried an alternative called FreeSwitch.

Turns out FreeSWITCH has a default setup where you can start testing/calling with some preconfigured standard accounts. Once I read some of FreeSWITCH’s Getting Started Guide (before moving to the more hands-on section named “Some stuff to try out!“) I ended up finding the accounts files and whatnot, to get my own set up up and running!

The “Some stuff to try out!” section is definitely a great way to get some SIP stuff up and running quickly and confirming if your setup is on the right track.

I know this is a long post :P

I wanted to kind of consolidate the stuff I learned in the process - as well as document it somewhere in case I forget and end up having to set up these phones from scratch again. Perhaps it might help someone else out there who might want to try a small intercom-like network for themselves. :)

#cisco phone #SIP #intercom #DIY #FreeSWITCH #Cisco 79xx #Tftpd #FYIT #telephone

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eprocurenet1 · 5 years ago

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Recommendations For A Small Office PBX System

You are exploring choices for a PBX framework to cover around 10 lines or somewhere in the vicinity. You PREFER not to depend on VOIP for outbound associations, however VOIP in the workplace itself is presumably alright.. what's more, perhaps for office-to-remote office. Your center (emulating your supervisors orders) is on simplicity of arrangement, reasonableness, and unwavering quality of the framework (the typical entirely evident administration bearing). FreePBX Cloud

Now...where do you go from here?

To go with a reason based answer for a little outfit probably won't be the be the best alternative. The explanation I state that is it isn't really a simple framework to oversee, particularly when you are thinking about connecting remote clients for an "on net" appearance. There are arrangements accessible for your size endeavor, however from my experience they aren't the best.

A redistributed arrangement will give you a similar look and feel, in addition to significantly more, of an exceptionally top of the line PBX type arrangement. It likewise makes the association of remote clients a lot simpler. In many facilitated, or redistributed arrangements, the requirement for VPN is disposed of, which can be hard to keep up for voice. Little organizations can work a lot of like enormous endeavors with a basic, redistributed media transmission administration.

Contingent upon your degree of solace with arrange arrangement and the executives, in the event that you mean to introduce and keep up the framework yourself, I recommend the sellers talked about underneath. When in doubt.....get help from an expert who has involvement in IP/VoIP needs investigation and stage determination for little to medium sized organizations.

You can keep up your simple POTS lines or whatever association with the PSTN you at present use, there is no motivation to relinquish that. Most IP based frameworks nowadays let you make a blended dialplan, where you can enhance your customary lines with a couple VoIP lines (SIP Trunks) that can be utilized for LD or International calls. Avaya Cloud

There are numerous variations of Asterisk that consolidate a realistic UI that includes most managerial errands like setting up your trunks(lines), making augmentations, enrolling IP telephones, setting up auto specialists and horde different choices.

For Asterisk based frameworks, I would propose looking at:

trixbox.com

asterisknow.com

switchvox.com

fonality.com

With any of these, your base needs would be the PBX, which is regularly a rackmount or midtower server with genuinely humble specs (Intel Xeon CPU, 1GB RAM, single or double SATA hard drives on the off chance that you need RAID, and on the off chance that you have (8) simple telephone lines, you would require a 8FXO TDM Card coordinated into your PBX case.

The main issue with a reference mark based arrangement is that it requires a great deal of work on your part. Unquestionably do some exploration on some facilitated or oversaw PBX sellers that administration your territory. Something on reason will likely not bode well.

On the off chance that you are searching for a greater amount of an apparatus type arrangement, I would look at talkswitch.com.

As a rule .... hope to pay $400-$600 per seat for the PBX, telephones, and maybe an oversaw ethernet switch.

Likewise, having your interior LAN arrangement for VoIP is significant. You need to execute an interior QoS (Quality of Service) instrument, regularly a VLAN that sections your IP Phones from your typical transmission capacity, so you dispense reasonable data transfer capacity for the VoIP.

I would not limit voip for outbound associations. So ideally your "Like" isn't an unshakable position...and you're available to outbound VoIP. There are a couple of good oversaw voip suppliers out there. Dealing with your own PBX is anything but a straightforward errand. You have to comprehend dial plans, did/dod, voice message incorporation. In the event that you need to do it well, you will need to have in any event a devoted person.... if not group. For 10 lines, it would almost certainly be needless excess. Switchvox Cloud

Simply my supposition, your mileage may change.

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asteriskservice-blog1 · 6 years ago

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Custom Asterisk Development Solutions in Australia For Businesses and Service Sectors

The Australian economy and GDP show a healthy growth trend with support from government and industries such as shipbuilding, mining, insurance, aviation and IT are growing apace. Communications forms the backbone of any business and the Asterisk platform is the best for small to enterprise grade VoIP communication applications despite the rise of Freeswitch. It is open source and has an active community of developers contributing to its evolution. Asterisk has modules that can be customized and put together to develop tailored communications systems that align with a user’s needs. It can be compared to Lego in which how you put together the bricks determines the shape you achieve. In the same way custom asterisk development by experienced experts makes all the difference to your communications platform.

Custom Asterisk development areas

As an open source platform with a huge library of modules and contributions from an active community, Asterisk offers standard options and customizations in various areas such as system, software, applications and modules in addition to incorporating external features like WebRTC through APIs. For instance, in the case of an application, all modules in a file system will load and slow down operations. Experts in Asterisk development configure and define relevant modules to be loaded in context which immeasurably speeds up operations and reduces load. Channel implementation is another area where extreme customization is possible due to experience and thorough knowledge of the features, desired outcomes and adoption of right channel technologies for super smooth call operations. Similarly, dialplan is extensive in Asterisk and are loosely defined. It needs an expert to implement internal APIs and include scripting for sound performance. Another area where knowledge, expertise and experience come together in custom Asterisk implementation is codec translations and negotiations. Given the variety of codecs in use transcoding plays a vital role in connection quality, clarity of audio and smooth flow. This is just touching the complexities of Asterisk where only custom Asterisk development can deliver solutions that hum along. This is especially evident when it comes to scaling a system involving multiple servers where a scalable communication framework implementation is necessary.

Representative Asterisk customizations

IVR: Virtually every other business implements an Asterisk based IVR but unless it is customized to be dynamic and configurable the IVR does not deliver expected results.

Dialplan programming: Crucial to inbound and outbound calls, dialplan programming needs to be customized to the user environment such as call center or broadcasting.

Call capacity enhancement: Asterisk does have a limitation in the way it can handle concurrent calls. Again, custom Asterisk development can provide workarounds to increase concurrent calls and call per seconds.

Choose a specialist in custom Asterisk development, one that has a thorough understanding of the critical architectural concepts and experience in delivering such solutions. It works out cheaper and better since they will have a lot of reusable modules that can be implemented fast for a tailored solution.

Source: https://voipconferencingsolutions.wordpress.com/2019/06/28/custom-asterisk-development-solutions-in-australia-for-businesses-and-service-sectors/

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bestasteriskdevelopmentcompany · 7 years ago

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Affordable Asterisk Support: Do You Really Need It? This Will Help You Decide!

Many of the large, medium and small enterprises use Asterisk which is an open source PBX solution. GVenture specializes in Asterisk Solution and helped many businesses from this open source PBX solution. For business, Asterisk is an affordable and highly powerful Linux based open source PBX solution. Asterisk solutions are now used by every large as well as small business organizations all over the world. GVenture is one of the providers who provide the Affordable Asterisk Solution.

Asterisks have a lot of useful functions that make it a strong IVR platform. Functions such as audio playback, digit collection, recording, database and web service access, optional speech recognition and synthesis, calendar integrations and much more are used in IVR. Through the Asterisk gateway or Dialplan languages IVR applications can be building and after that, they can integrate virtually with any external system.

What's Right about affordable Asterisk Support?

There is no need for additional hardware for Voice-over-IP in Asterisk. A single/multiple VOIP can be used for the calls. Asterisk provides Voicemail services with Call Conferencing, Directory, Call Queuing, and Interactive Voice Response which is the most important features of it. It has support for three-way calling, caller ID services. Affordable Asterisk Support is provided by GVenture which will be very helpful to you in every step.

GVenture is the best IVR solution Provider!

IVR aka Interactive Voice Response is a telephony technology in which a touch-tone telephone is to be used to interact with a database to acquire information. IVR technology doesn’t require the human interaction at all over the telephone. The user access will be only allowed by the IVR system if it is predetermined. GVenture will help you in providing the latest solution which you want to be implemented in your business IVR.

Mostly large corporate PBX’s today uses IVR. Naturally, it is automated voice menus which you’ve heard while talking to your telecom operator, bank or any insurance company. The recorded voice asks you for an input which wills supposedly for the transaction. You can input the response in the form of digits (DTMF or dual tone multi frequency tones. Then after the transactions which will be executed will be resulted what the user has given the inputs. We are going to inspect IVR coding so the hardware configuration needed. Though there is a lot of IVR answer provider we guarantee you GVenture crack the list.

Full 24*7 Technical Supports

If finally, you have decided to implement Asterisk Solutions in your business we suggest you go for GVenture’s Asterisk Solution in your business. The feature of which will enhance your telephony experience and will fulfil your basic and advance communication needs and also you can customize it as per your business requirements. These all are the signs of success in your business. Best IVR solution provider is the GVenture.

About GVenture, IVR solution Provider

GVenture has a fully fledged and expert team of Asterisk Developers as well as Affordable Asterisk Support professionals. Hence along with the required customization and implementation of Asterisk Solution at your premises, we also provide 24*7 full and live Technical Support so that all of your queries and issues are resolved on time so that you can run your business without any interruption and delay. Our relationship will be valued so that we provide excellent customer service.

#ivr solution provider #affordable asterisk support #asterisk solution #gventure technology

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