VoIP Jobs

Hello Freelancers, We are UK Based Company. We have leads we want someone to convert those leads into sales. You have to be professional English speaker if you want this job. We offer CCNA course in UK our next batch will start from 4th of November. Please write "1pv6" in your bid so i can filter automated bids. We have script you just have to call the persons and inform them about our...

I need a frontend with asterisk api to open the client's registry when the connection is answered, this connection is generated in principle from a predictive dialer inside the asterisk but can be dialed by the operator also in order to return a contact, and that system you must have the option to import a CSV file of mailing that we receive to dial. after that I need a backend to follow the ...

Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

We are looking for campaigns that need outbound or inbound services. We do cold calling, lead generations and appointment setting.

inside sales agent for real estate prospecting calls

I have fully and complettly configured Asterisk, but I need someone to configure on my site sipjs to works with my Asterisk

Position is responsible for placing outbound calls to employers/plan administrators (health plans) who have not provided a complete response. Position does not receive inbound calls. About 1.5 weeks of initial training time.

Dear all, I am seeking a pfSense expert to point us and tutor us on how to accomplish the following: - We have 3 Broadband lines, later on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 -...

So, I have a stock standard, fresh FreePBX installation (in production) I also have a ASP.NET project that has coding in it that pulls call information from the Asterisk database. The ASP app is currently being rebuilt and relaunched and so help is needed with someone in experience in all of the areas mentioned in the project title to put the pieces back together. This project is broken down in...

Looking for one time setup of auto dialer set up on server.

Install and configure FreePBX 14 on Digital ocean and setup, needs to support commercial modules...

Specifically, I would like to have knowledge on: 1. Cisco 3925E router with 3 VDSL lines (a) to load balacing b) voip 2. Cisco 6509E a) Update IOS and Module b) Configuration of the local lan for several servers 3. Cisco WS-SBS 3032 (Blade Switch) 4. Cisco WS-SBS 3130G-S (Blade Switch) 5. Cisco ws-c 3560-48PS-S I want someone who can set them up. Thank you

Hello. We are looking for Voice Broadcasting System. Contact us if you have ready and can show us demo instantly. Thank you.

Hi. we need part time permanent remote freelancer who can work on our existing project to fix bugs, add features etc. this app is built for E-learning one on one audio, video sessions just like skype. this app has: audio, video, screen sharing, conferencing , ios native and android native apps. it is built on webRTC, Node.js , angular.js , asterisk, xcode, callkit, android studio. we need som...

Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDrive App. Need check the requirements from Pipedrive. [login untuk melihat URL] The Features I need are: Click to Call (CTI) - Click in contact in PipeDrive and call to customer. Call History - All calls made will be logged in the customer details of Pipedrive. Have to match the phone number from Freepbx int...

asterisk person to work on an existing product to fix and update the product knowledge needed: PHP Asterisk database

We have iPECS-100 behind Chekpoint firewall. All customers use soft phones clients. Have periodically experience communication problems. Need a person for a one-time or permanent job

well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should ...

Can you create a feature so that my users call face time each other on my website? #VOIP #video

System requirements Registration screen Logon screen online Users screen Server php and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple appli...

Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de llamadas necesitamos...

i need help in my freepbx i m not getting call waitng options

Only for gurus familiar with Bitrix who can setp Bitrix application to work with our PBX Need to use Telephony REST api

a2billing setup, calling card, PBX, Auto dialing, SMS

I would like someone to config a Cisco IAD 2435 for MGCP. I dont want to use Cisco call manager. I can give you remote access to this device for testing.

I'm stating voip system and calling card with soft-phone service. I want some one who can install freeswitch with fusionpbx billing solution.

i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

Configurar um server opensips/kamailio para controle de ddrs,chamadas para gateways externos , chamadas entre ramais , cdr com interface grafica.

Freepbx installation and configuration over raspberry Integration with SuiteCRM: popup contact form with incoming call, clicktocall for outbound calls ann log incoming and outgoing calls SuiteCRM is in local server

i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

Hello. We are Looking for Call Center Solution based on Freeswitch (Multitenant) (Not Vicidial or GoAutodial) along with Source Code. If anyone have ready, contact us and provide us demo. We will purchase instantly. For more information you can chat with us. Thank you

we need a phone system based on twilio platform.

I need an android application for video streaming. (Bigo live, [login untuk melihat URL] kind of app). Please review specifications document before bidding.

Ringless voice mail broadcast service. Automatically delivers your pre-recorded voice message straight into consumers’ voicemail, without ringing the telephone. *** Replicate the voicecasting capabilities found here - [login untuk melihat URL] called voicecasting*** Voicelogic has a service that allows you to make a call that will go straight to voicemail without ringing the customer. I wa...

Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined Deadline: 4 weeks

Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets