Freelance Jobs

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Asterisk PBX jobs

Elastix IVR

Aug 20

Requires Asterisk PBX, PHP, Website Design knowledge.

We are a telecommunications company based in the UK. We have recently started to broach into the corporate side of the IP market after many years of succsessfull trading on the TDM markets. We have decided to start swapping out customers from ISDN onto our own IP network.

I am looking for a highly motivated person to help us design a new corporate service using asterisk and elastix.

It will need to be fully customized and offer new products as well as ones already available within elastix. We are currently open to suggestion on the billing aspect but a2billing seems a resonable choice as its bundled with elastix.

We are also open to suggestions when it comes down to using elastix, if you are aware of a more stable solution we are more than happy to listen. Our company owns its own datacentre which all this will be hosted inside of.

We are anticipating high demand, so the solution will need to be fully expandable, and grow as demand grows. The concerns I will need meet are , Linux Security, Very Accurate Billing, Nice polished interfaces for clients, and regular statistics and billing.

We are also looking at a new Video Conferanceing solution to encoporate into the packages. This is a long term arrangment as we have a lot more scope for future development on other systems we have in the pipeline.

I would also like to point out, we are not looking for developers based in India and would prefer sutible people within the UK.

All works carried out will be owned by the company, and we require strict revision documents, as well as any documents relating to its function and how to avert disaster.

Please contact me on the board, and make sure you attach a CV for our viewing. Once a person has been chosen we would like to have a phone discussion on how to proceed. Many thanks.

Nemesis Data

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VOIP SIP Trunk Provider Research

Aug 17

Requires Asterisk PBX knowledge.

I am looking for outbound VOIP SIP Trunk providers. The pricing plan of our current outbound SIP provider we are using when

calling within the USA is (any state in the US calling any other state in the US – excluding Hawaii):

3 cents/minute
30 second minimum call duration
6 second call increments
No monthly fee
Unlimited simultaneous lines

We are looking for more competitive outbound SIP Trunk providers. Please note: We are not looking merely for VOIP SIP phone line/extension service providers where we have to purchase X number of lines. Instead, we are looking for a SIP Trunk provider so that we pay a per minute cost and can have either 1, 10, 100, or any number of phone lines making simultaneous calls dependent on need.

What we are looking for (the ranges are below):

* less than or equal to 4 cent/minute
* less than or equal to 16 second minimum call duration
* less than or equal to 6 second call increments
* No monthly fee or subscription fee
* Unlimited simultaneous lines (or an extremely high limit… "extremely high" being 1,000 or more)

Of course we would rather aim lower than higher for the first three of the five criteria listed above. The deliverables of this research project are to provide us with the contact details of a reputable outbound SIP Trunk provider that offers the lowest per-minute rate (within US) along with the lowest minimum call duration (per call) and lowest minimum call increments with no monthly fees/subscription fees and preferably unlimited simultaneous lines available.

The voice quality must also be good from the proposed service provider.

We may choose multiple winners for this Project. When you have found a SIP Trunk Provider that matches our requirements, please message me letting me know the criteria above for each service provider proposed. For example, if you found an outbound SIP Trunk service provider offering a 10 second minimum call duration, 6 second increment, 2.5 cent/minute rate, and unlimited simultaneous lines please email me with this data. If you have found more than one, please provide details associated with the additional service providers as well.

The pricing for this research project should be based on you being able to provide us with one qualified service provider. Upon being selected as a winner of the Project, payment associated with the research will be funded in escrow. You will then reveal the service providers contact information (for the first service provider only, if you have several), and we will contact the service provider and test the voice quality. Upon passing the requirements, we will release escrow. If you have more than one potential service provider, we may fund escrow again for the same payment amount, and upon your releasing the information and us testing, we will release escrow.

Please call to confirm the legitimacy of your rates prior to messaging me. Also please confirm that they are offering SIP trunk versus just a SIP phone line/extension. Thank you!

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Telephone Answering Service Application For Asterisk

Aug 17

Requires AJAX, Asterisk PBX, PHP, Software Architecture knowledge.

In need of a programmer who has experience writing programs that use the Microsoft Windows API Telephony Application Programming Interface (TAPI) & SIP.

We are interested in having an answering service call handling and scripting software developed identical to www.nsolve.com but with added features that is designed to work with asterisk

Look over www.nsolve.com to get a feel for the functions of the application.
Application must include billing module as well as a scripting module where representative can build questions and answers to questions within user profile

Application will be used in conjunction with a Fonality asterisk pbx The systems gui is hosted so user must be knowledgeable of creating an application that can run in conjunction with the hosted gui. Your suggestions are most welcomed to derive at the most suitable solution

The program should run on windows XP, Vista and Windows 7 environment.

We would prefer C++ with a nice GUI.

I will provide a detail list of features the application should include at the beginning of the project.

Freelancer should be VERY PROFICIENT WITH ASTERISK AND WINDOWS TAPI APPLICATIONS

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NEW INSTALLATION FreePBX & A2BILLING Server

Aug 14

Requires Asterisk PBX knowledge.

We are looking for consultant you has experience in successful remote installation on Asterisk Server FreePBX and A2Billing. Please send email and price to with a contact phone number PLEASE DON NOT reply if you have NOT done more that 25 remote installations!

OS: CentOS 5

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Call Center Application Using Asterisk

Aug 13

Requires Asterisk PBX, PHP, Software Architecture knowledge.

I am new to Asterisk and need some help getting to where I need to be. I am opening a callcenter and I will want to use softphones. I am running AsteriskWin32 and would prefer if you can maintain me on Windows. Im not a fan of Linux. Right now, I have 10 SIP trunks, this will grow. I also have a broadband connection of 4 mb. I can make and receive calls at the moment, but thats about it. Also, I have the one computer that I will use, but I will be expanding. This computer is serving as the Asterisk server and the first operator, but as we expand so can the system.

What I need (or so I think): I need a pbx manager that will manage incoming calls. When an incoming call comes in, I will want the app to open a webpage and transmit the caller id and dnis info in the page (example: www.mydomain.com/callcenter.asp?dnis="dnis from caller"&callid="caller id from caller").

Also, depending on the dnis, I would like a welcome message (hold music) to play before the call is answered by an operator. What I mean is, depending on the dnis info, the hold music will be different.

In addition to this key feature, I would like other features like call transfer, hold music (determined by dnis), call recording, search for available operator, call forwarding to specific phone numbers defined by dnis, reporting (like queue, call length, various measurements), sending incoming calls to remote locations, and outgoing calls via skype when we hit a max number of minutes or when we call long distance. Chances there will be more/change orders as we go forward, so you will most likely make more money in addition to your first bid.

I also need a soft phone, I am testing a free version of a few, but if you like one, let me know.

Please keep in mind that we will be expanding, so I dont want to hit extension limitations.

IMPORTANT: part of your task will be to educate my as to the use of this system. I am an asp programmer so hopefully I will be able to keep up. If it works out, you may become a regular contractor for me for system maintanance.

Finally, I have a beginners budget for this project, so please give me your best bid.

Any questions, let me know. Thanks

Jack

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Asterisk And Vicidial Installation And Customisation

Aug 13

Requires Asterisk PBX, Linux knowledge.

We are looking for consultant who has done 20+ remote installations and customisations of Asterisk and Vicidial.

Please send email, price and previous experiences. We require intallation and customisation of Asterisk and Vicidial.

PLEASE DO NOT reply if you have NOT done more that 20+ remote installations!

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Asterix/A2billing PBX Installation For Calling Cards(Urgent)

Aug 12

Requires Asterisk PBX knowledge.

Immediate/Urgent requirement for Asterix/Billing software installation and configuration on 4 separate servers for calling card business, PBX etc…..

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Full Configuration Of VoipSwitch Server

Aug 12

Requires Asterisk PBX knowledge.

Urgent requirement for a voipswitch expert to assist with the configuration of all VOIPSWITCH modules including (calling cards, reseller module for calling cards, DID, residential/Business phone, Mobile VoiP) .

This is very urgent!!!

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Setup Business Virtual Pbx

Aug 12

Requires Asterisk PBX, Linux knowledge.

Hello.,

project very simple, we have a voip account and a linux box (currently UBUNTU installed, but we will install whatever OS you recommend) , we need to setup a pbx for our business and mobile staff with automated attendant, follow-me and forwarding features.
by using freepbx or any other FREE scripts to integrate our voip account with the pbx system.

something similar to www.grasshopper.com, but only for our business where callers get greeted by automated attendant and then gets forwarded to the person or department they wish to speak to.
besides whats mentioned above, the following features are needed:
- unlimited extensions
- unlimited voicemail accounts
- follow me feature
- call forwarding
- live call transfer feature
- automated attendant
- name directory
- music on hold
- webbased administrator access, where we can add, edit pbx functions and extensions
- online access for staff members to their mailboxes to listen to thier messages

and the rest of grasshopper features, EXCEPT the virtual fax we dont need

terms:
- look at grasshopper site, if you think you can do it then bid
- if you havent done any pbx and voip project, your bid will be rejected
- we only will pay by papal
- no advance payments
- we will provide remote access to our linux box to work directly there

P.S. this is just a start, we already have more pbx setup projects and voip projects for our different divisions and customers

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Trixbox Queue Monitoring Webpage For Iframe (for Meral)

Aug 11

Requires Asterisk PBX knowledge.

We need a webpage on our Trixbox/FreePBX box that will display queue statistics in a easy to read format.

Current requirement is to only show:
"Number of Calls on Hold"

although later we may wan tto include:
"Max limit"
"Average holdtime"
"Member calls answered stats"
"List of CIDs of calls in Queue"

Add these but make them selectable via the URL for the page. Eg:
?queue=600&average=y&CIDlist=y

Page only to create each time page is called.

Page URL should have a security string.

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Web Application For Sending Messages By Phone.

Aug 11

Requires Asterisk PBX, Java, PHP knowledge.

We are looking for a professional / company to develop web application.

This Web application is to create an audio message and send it by phone to a programmed number or numbers, the service is enabled by a key number which will allow in a limited number of messages, each message sent deducted from the total, if have been exhausted may no longer be sending more message.

Detailed project description:

1.The user logs if it is the first time is essential to register at least its full name, sex, date and place of birth, email and password. If the user has been registered may login with its email and password.

2. Upon entering the application, the user has a history of messages sent and the balance to send new messages, if its balance is exhausted it can enter a new key and can send messages again.

3. To program a new message, the user capture it in a text field or select it in the message history.

4. The instructions are converted into an audio file (this file should be a maximum of 10 seconds, this period will be configurable through the application administration).

5. The user selects the telephone numbers where message is sent, the date and time of delivery and shipping policies, they are, the order of call, if confirmation is required, etc.

6. On reaching the scheduled time the message is sent which is preceded by a sponsor message 10 seconds (audio messages already contained in the application), the application must provide the elements for the management of these messages. To select "sponsor message" to send with "user message", words in the field text of the "user message" determine wich "sponsor message" is sent.
Example: Sponsor1.mp3, keywords: "restaurant", "lunch", Sponsor2.mp3, keywords: "travel", "hotel", "plane" User Message "Schedule lunch with family at 3 pm ", this message would be preceded by sponsor1.mp3 message (lunch is the keyword in this case). User Message:"My plane leaves at 9 am", this message would be preceded by sponsor2.mp3 message (plane is the keyword in this case). As mentioned above, the application must have the elements to Administer this feature.

7. The application has the option of storing text fields that the user only want to store and doesn´t want to send. The program is a software solution that can generate audio messages from text messages and send them through VOIP. The communication application utility is similar to Autoreminder program (voicent.com) .

It has an estimated 50,000 users. The average number of messages per user is three per day so it would have 150.000 messages per day. It is considered an average of 12 peak hour service so you would have 12.500 calls per hour. We consider send messages on a range of 15 minutes to simplify infraestructure, and considering that the message would last 30 seconds (10 sec. In advertising "sponsor message", 10 sec. "user Message" and 10 seconds of system messages) and takes 30 seconds to make the call, a total of 1 minute per call, which requires 834 lines (each line would make 15 calls per hour, 180 day to 60 phones different line takes three hours).

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Configure "Follow Me" In FreePBX

Aug 10

Requires Asterisk PBX knowledge.

Im running FreePbx, A2Billing with a custom dial plan. Calls come into the system and then are forwarded to outside extensions (mobile phones) using follow me. That works fine.

What I need is someone to enable the extension owners to turn their Follow Me settings off and on.

I need the owner of each extension to be able to call into the system using a common DID.
I need the system to authenticate them by prompting the extension owner to enter their extension and a 4 or 5 digit PIN.
After authentication the system will give the extension owner 1 of 2 options
1. If their Follow Me settings are ON, the system will prompt them to press a code to turn their Follow Me off.
2. If their Follow Me settings are OFF, the system will prompt them to press a code to turn their Follow Me on.

When the extension owners dial into the system, that should be the only choice they have. They should not have access to any of the other feature codes and they should not be able to do anything else other than turn Follow Me on or off.

When the ADMIN looks at each extensions Follow Me settings in the FreePBX dashboard it should reflect if that extensions Follow Me is enabled or disabled.

If you do not have favorable feedback dealing with Asterisk or FreePBX please dont bid.

This job will be awarded the first week of September.
Payment upon completion.
I would hope the winning bidder would have the job complete within a day or two at the most.

If you have any questions feel free post them.
Thanks

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Voip BN PROJECT

Aug 10

Requires Asterisk PBX, PHP knowledge.

C++ developer. voip and sip applications. ability to integrate voip with internet browser plugin. Softphone implementation into a internet browser and PHP WEBSITE. If you qualify for this send me an email with your resume and experience and any work you have done in the past if posible.

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A2Billing, Asterisk, FreePBX On CentOS VPS For WISP

Aug 9

Requires Asterisk PBX, PHP knowledge.

As a small WISP (wireless internet service provider) we would like to start offering VoIP to our customers. We have tried the billing systems and SIP registrars of offsite hosted ITSPs, but it does not work for us.

We want to install Asterisk + FreePBX + A2Billing on our own hardware running XenServer 5.6 and CentOS 5. It has two network interfaces (eth0 to the internet with public IPs and eth1 to our internal network with private IPs). We need to set it up this way to avoid NAT problems experienced with SIP.

We are currently making use of a single ITSP giving us SIP to their side with multiple DIDs for incoming calls, which we will in turn assign to specific customers.

We need to be able to configure our own billing/rates (different for business and after-hours) to bill customers. Monthly "line rental" should also be taken into account.

Customers should be able to register to our server using SIP or IAX.

We will do our own customised branding if you can show us where the images and HTML files for A2Billing are located.

Basic training/instructions on where to look when something goes wrong would be essential.

A basic backup of MySQL, configs and HTML files should be set up on a cron job.

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Setup Asterisk Imap Voicemail

Aug 9

Requires Asterisk PBX knowledge.

Need someone to setup imap on an asterisk server and document process.

Server – centos 5.4 32bit
Asterisk 1.4.29

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VOIP Support

Aug 9

Requires Asterisk PBX knowledge.

We have a VOIP system within our company which we are experiencing numerous technical difficulties.

We require someone to examine our system make recommendations & a cost to fix so it is working efficiently.

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Loking For Somone In Fiji Islands/Australia To Host Voip Gat

Aug 8

Requires Asterisk PBX knowledge.

Lokking for someone whio can configure a skype prettymay call center or aestrix voip gateway to forward calls from us based people who want to call fiji , person could work on monthly payment or one time fees if he/she can provide reliable service,
Looking for people both in Australia and fiji and ghana

The person should be familiar with skype or aextrix server and should be able to provide a voip gateway and voip call forwarding from these contries to local At the moment i have a max budget of only $100 so please do not bid more than that.

Thanks

Ajay

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Custom Inbound Trixbox

Aug 7

Requires Asterisk PBX knowledge.

If the incoming call is one of the numbers send them to custom-screen. Ideally I would like to send the call to a group but it is currently not working. I am using Trixbox.

[custom-incoming]
exten => s,1,Answer
exten => s,n,AGI(calleridname.agi)
exten => s,n,GotoIf($["${CALLERID(num)}" = "XXXXXXXXXX"]?custom-screen,s,1)
exten => s,n,Goto(ivr-2,s,1)
exten => s,n,Hangup

[cusom-screen]
exten => s,1,Wait(1)
exten => s,n,Dial(SIP/7045,30,m)
;exten => s,n,Goto(ext-group,9900,1) ring grp to ring to private no agents
exten => s,n,Hangup

Thanks
Sam

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Sugarcrm And Asterisk

Aug 6

Requires Asterisk PBX, PHP knowledge.

Need to do SugarCRM integration with Asterisk-based dialer.
Only who has the experience and can allocate time should bid.

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FreePBX/Polycom Configuration File Creator

Aug 6

Requires Asterisk PBX, Linux knowledge.

Wed like the app to automatically generate configuration files that are stored on a separate ftp server (pure-ftp), based on Extensions being created in FreePBX. We are planning on utilizing the AccountCode field in FreePBX for this application. Example:

We create extension 100 in FreePBX. The AccountCode for that extensions is P601:0004f21AAAA:T-7,DY,Y9. P601 represents the phone type in this case Polycom IP 601. The 0004f21AAAA.is the MAC Address for the phone. T-7 designates the GMT offset is -7. D represents Daylight savings time values: DY yes, or DN no, 9 represents whether the extension has to dial 9 to get an outside line value: 9Y or 9N

The program is launched when an extension is created or modified. The program uses the data from the asterisk database and a file called server.cfg located in /usr/local/bin to generate five files in a specific directory on the ftp server; MAC.cfg, MAC-user.cfg, MAC-dialplan.cfg, MAC-custom.cfg, MAC-server.cfg

Server.cfg only has two variables. Server=IP Address1, Proxy=IP Address2 (Wed prefer to store these values in FreePBX if anyone can think of a good location)

The files location is based on the first four characters of the Account code. For example if the Account Code begins with P601 the files are located in /home/ftp/P601. If the Account Code begins with P550 the files are located in /home/ftp/P550.

The following are examples of the different files to be created. Each file should be create and or/overwritten each time the program is run, with the exception of MAC-custom.cfg. This file needs to be created, but if it exists it must not be over-written.

MAC.cfg
<?xml version="1.0" standalone="yes"?>
<!– $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ –>
<APPLICATION APP_FILE_PATH="sip.ld"
CONFIG_FILES="MAC-custom.cfg, MAC-user.cfg, MAC-dialplan.cfg, Mac-server.cfg, sip.cfg"
MISC_FILES="" LOG_FILE_DIRECTORY="/logs" OVERRIDES_DIRECTORY="/overrides" CONTACTS_DIRECTORY="/direcory"/>

MAC-user.cfg
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<reginfo>
<reg
reg.1.displayName="$Extension" (pulled from FreePBX database)
reg.1.address="$Extension"
reg.1.label="$Extension"
reg.1.auth.userId="$Extension"
reg.1.auth.password="$Secret" (pulled from FreePBX database
reg.1.lineKeys="1"
/>
<SNTP
tcpIpApp.sntp.daylightSavings.enable="1" (value 1 if DY, 0 if DN)
tcpIpApp.sntp.address="tick.ucla.edu"
tcpIpApp.sntp.gmtOffset="-28800" (value = T value times 3600)
/>
<msg msg.bypassInstantMessage="1">
<mwi msg.mwi.1.subscribe="3001" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*970"/>
</msg>
OTHER STANDARD TEXT
</reginfo>

MAC-dialplan.cfg (varies on 9 variable)

<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<!– SIP Application Configuration File –>
<!– $RCSfile: sip.cfg,v $ $Revision: 1.483.2.30 $ –>
<SIP>
<dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1" dialplan.applyToUserSend="1" dialplan.applyToUserDial="1" dialplan.applyToCallListDial="0" dialplan.applyToDirectoryDial="0">
<digitmap dialplan.digitmap="[1-8]xxx|[9][2-9]11|[9]011xxx.T|[*]xxx.T|9[1]xxxxxxxxxX|9[2-9]xxxxxx" dialplan.digitmap.timeOut="3|3|3|3|3|3"/>
</dialplan>
</SIP

MAC-server.cfg
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<localcfg>
<server voIpProt.server.1.address="Server"/> Based on Server.cfg
<SIP>
<outboundProxy voIpProt.SIP.outboundProxy.address="Proxy"/> Based on Server.cfg
</SIP>
</localcfg>

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Asterisk CDR Report

Aug 5

Requires Asterisk PBX, PHP knowledge.

I need PHP script which which extract the following details from my Asterisk CDR for any given DID which is being stored in `userfield`.

> Date-Time
> DID (I am recording this is `userfield`)
> CLID
> t1 (time when call was received)
> t2 (time when call was answered by agent)
> t3 (time when call was hung-up)
> t4 (Hold time: t2-t1)
> t5 (Talk time: t3-t2)

Fancy HTML etc is not required. Just simple tabulation will do.

Call Flow:
==========
[caller] -> [Trixbox IVR] -> [Trixbox Call Queue] -> [Trixbox Human Operator]

I can send a sample of about 100-200 records from my CDR.

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VICIDIAL Report Customization

Aug 5

Requires Asterisk PBX, Linux, PHP knowledge.

Hi,
We are using Vicidial 2.2.1. We need development and customization. This project is for customization of existing Vicidial Report.

There are two reports namely;
- Agent Performance Report
- Inbound Report

Currently, the Date Input method allows you to choose the From Date and To Date.

Requirement: I need to make these reports in Date and Time Range instead of Date only.
e.g. From 27-10-2010 10:00:00 to 27-10-2010 13:00:00

I can attach the Dump of php scripts, but i would prefer if you have worked on Vicidial or have a running system to make available neccesary files and test the same.

Thanks
Ruchir

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Asterisk , VOIP

Aug 5

Requires .NET, Asterisk PBX knowledge.

We are a VOIP provider and we need solutions for our existing issues in the Asterisk server.

"Only serious bidders can bid- Please do not spam with recycled application"

"Experience in both Asterisk and A2Billing is a must"

The job description is given below, however, the first two points will be done in this bid, and on successful completion , the other points will be done.

1) The asterisk server should be checked for the registration fail of the phones
2) The dial plan should be reviewed for the disconnection of the phones at times and unsuccessful call completion
3) New Dial plan should be written for the usage of multiple terminations for selected countries
4) A2Billing should be customized according to our need.
5) Chances are there to write web services which may talk to the application .

Optional expectation:
1) ability to program in .NET
2) Knowledge of A-Z Terminations
3) Knowledge of Calling Card solutions

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HOZAND

Aug 4

Requires Asterisk PBX, Computer Security knowledge.

I WANT A HYIP TEMPLATE HIGHLY UNIQUE WITH ADOBE FLASH

this template most be highly professional even more than ptv partners..

budject $30-$80

template should be completed in 24 hours
send me your Ym by message and i will add you so we can chat better…

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Data Correction

Aug 3

Requires Asterisk PBX, PHP knowledge.

Need Data Correction web application through an asterisk server,the old data to be imported through XlS or Sql into the system &amp; to appear to the agent like a 2 pages in a book the left is the original data, &amp; on the right the editable data , the right page wont be opened unless the agent called the customer &amp; verified with him his ID for example
after verification the agent edits the data &amp; categorize the call as a one of many categories
the call is recorded through the Asterisk server &amp;need to be attached to the page of the client with its call category
so later we can get as much reports as we can from these categories
on the users side we need each system feature can be either allowed or denied for each user
live counters need to be on a main screen for managing the project
other counters needs to be for the each user so he/she can know how many calls he did &amp; under which category
many details will be explained later
thanks

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Asterisk Answering Machine Detection Settings

Aug 2

Requires Asterisk PBX, Linux knowledge.

Hi,

We need settings for Asterisk for detecting Answering machines for US numbers. Detection should be more than 80% accurate and should work for every type of Answering machine. Nothing additional needs to be developed. We just needs Asterisk AMD settings.

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Custom Caller ID Based On Number Dialed Asterisk

Aug 2

Requires Asterisk PBX, Linux knowledge.

We have a client that owns a couple flower shops throughout the country. The use Asterisk 1.4.25 at their headquarters. At the headquarters they have a small call center (4 agents) that follow up calls makes calls on behalf of their remote stores. They want to their phone system to change the Outbound Caller ID based on what phone number the call center agent dials, if that number exists in their customer database. Heres an example

Agent = Jane Smith, Exten = 100, Default Outbound Caller ID = 310-555-1212
Customer1 = Joe King, Phone Number 661-555-1212, Local Store Phone Number = 661-888-1212
Customer2 = Diane Wallace, Phone Number 212-555-1212, Local Store Phone Number = 212-888-1212

Jane Calls a Joe King
Asterisk looks to see if Joes number is in the customer Database
If it is, then Janes caller ID is changed to 661-888-1212
If not then Janes caller ID stays 310-555-1212

Jane Calls Diane Wallace
Asterisk looks to see if Dianes number is in the customer Database
If it is, then Janes caller ID is changed to 212-888-1212
If not then Janes caller ID stays 310-555-1212

We can run a dump from the customer database to create a csv file that has Customer phone number, Store Phone number. Well update the csv file nightly. But it will always be in the exact same location on the asterisk server and the format will never change.

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Configuration Of Asterisk

Jul 29

Requires Asterisk PBX, Linux knowledge.

Hello!

The task is to configure the software ASTERISK to 5 internal lines inside the server through remote access.

The software must support all functions of call center.

We are interested in price and time of doing it.

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