VoIP Jobs

I am using vicidial and I need my callers to be able to enter survey information as they speak to the customer. I don't care which solution is used whether it be a simple template, google sheets, or jotforms. I just need someone who is competent and can solve this problem. [login to view URL]

I am using vicidial and I need my callers to be able to enter survey information as they speak to the customer. I don't care which solution is used whether it be a simple template, google sheets, or jotforms. I just need someone who is competent and can solve this problem. [login to view URL]

Set up cloud based Install of elastix v4 in AWS instance Setup Elastix 4 firewall, iptables, fail2ban, etc. Setup trunk and 5 extensions.

We require someone who has experience using / programming in Asterisk to provide a white labeled app for use on Iphone & Android. We require all features as noted here: [login to view URL] We also require call recording (to be stored on server or customers server) as well as Caller ID forwarding.

Need a simple Zello to SIP interface that will allow a phone on an Asterisk PBX to call into a Zello channel. When talking on the phone it will transmit/receive over the Zello channel.

We have a freepbx server, and we need to make calls based on csv files. We may can use our server, or maybe another third part service, if suggested/approved.

Good afternoon, Please note that the job posting is for *native Swedish speakers residing in Sweden only*. Applications from non-natives and non-Swedish residents will be disregarded. We are looking for Swedish natives to record short sentences on various general topics (retail, banking, telecommunications, etc.). You need to be physically located in Sweden in order to be able to participate in...

I am using vos3000 and I am looking to integrate it with whatsapp api.. Basically I want my Vos3000 to only pass voip calls whos destination numbers are registered on whatsapp platform. If a destination number doesn't exist on whatsapp then it is declined

Hello I need to setup the Bitrix24 telephony system. Attached to 3 different SIP servers. Thank you

60 to 70 percent work is already done on freepbx but something's still incomplete in my project. First i want to explain what is my requirements. I want a asterisk base caller/callee filter solution on PHP which can filter incoming traffic and create whitelist and black list base on condition of repeated callee/caller and non repeated callee/caller with VoIP call bandwidth optimizer. It wil...

I need a agent who can do 30 appointments a week for Australian process via cold calling. Data and VoIP would be provided and if you get 30 appointment every week I am ready to pay you 3500 - 4000Rs every week. Don't waste your bid if you don't want to work under this payout

The main part of a project is a creation of software which will help to manage GoIP equipment (Hybertone manufacturer - gateways and sim banks). All the features which will be included should be available in the interface - Some basic features are: 1:Simulation of Human behavior -Generation of the flow of incoming calls -The list of 'preferred' numbers. -Imitation of SIM card movement ...

need to configure my vos 3000 softswitch with my goip 32 and make script working for InterCall between ports in goip

hello i need to open PKG file and add some function on it its my goip gateway update but i need add on function

I have Isabel installed, i need the following 1. configure 1 number and 3 simultaneous calls? 2. configure power dialer and tell me how I can load my clients' contacts 3. manage predictive calls on any softphone

I have Isabel installed, i need the following 1. configure 1 number and 3 simultaneous calls? 2. configure power dialer and tell me how I can load my clients' contacts 3. manage predictive calls on any softphone

I need to connect an 8 channel DINSTAR gateway to my asterisk server and create extensions to configured and ip iphone.

need make serve sip and voip configuration

Looking to develop VoIP call forwarding Portal.

delay the buy signal on asterisk or opensips, when the call is hangup delay between 6 or 7 seconds to receive the bye

I need a call center that is willing to work COMISSION ONLY (PLUS MULTIPLIER BONUS). I have a script and I will be giving you the leads that you can dial out. You must have: YOUR OWN VICI DIAL or other CALL CENTER SOFTWARE (PREDICTIVE) Customer Service Experience Ability to read and write English ability to set appointments if clients are busy *I WILL PROVIDE LEADS* MUST WORK THESE HOURS: ...

Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Option to add delete sound files. 6. Branding Thanks Vidhya Sagar Dixit

Hello, I am using Issabel 4 with asterisk 16. At the moment I can only upload audio file at 8000 hz . I would like to know if it's possible to use .wav files at an higher rates like 16000 hz or 48000 hz in order to increase audio quality. Audio files should be used for all the audio parts: IVR, Announcements etc...

Freeswitch Expert required for some outgoing projects/scope.

Looking for someone experienced in mainly Mirta PBX or Asterisk by itself is less preferred but might be an option.. you would likely be a voice Engineer and will need to train people in use of and maintenance the PBX.

i need to provision a cisco phone into 3cx

Hi, I need a windows pc softphone with following features. 1. SIP server information will be configurable through an API. 2. balance display through api. 3. Auto answer the incoming call 4. Play a sound file to other party after answering the call. 5. Branding Thanks Vidhya Sagar Dixit

i need someone with past experience with ozekisms software

we need to fix some issues in cisco call manager like add fax direct number etc

I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.

I'm using freeswitch and want to configure mod_gsmopen so calls can go via sim

install ct3504 controller and 43 Access points . the configuartion has to be done remotely .

Servidor VOIP para integrar con aplicaciones moviles