Asterisk PBX Jobs
Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
Hire Asterisk PBX DevelopersI 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.
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.
Hello. We are looking for Voice Broadcasting System. Contact us if you have ready and can show us demo instantly. Thank you.
Our project consists of two parts: 1) Select the correct customer inside POS automatically whenever the customer makes a phonecall to our call center. If the phone does not exist on the database (both mobile and phone partner fields) it should automatically open the add new customer of POS. The functionality is similar to this module: [login to view URL] but inside POS environment 2) An interface ...
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...
I have a company with multiple phones and would like to keep better track of conversations. I need a simple system that when called redirects to my company numbers. I also need it to record all conversations also.
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 to view 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 into Pipe...
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 ...
I am looking for an off-the-shelf voice messaging platform like [login to view URL] has developed.
Can you create a feature so that my users call face time each other on my website? #VOIP #video
I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be completed in order for push notific...
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
Hi, I have Php house made CRM, want to integrated 3CX PBX VoIP with CRM, Adding extra field in the customer profile (email) and generate report. More information are available in the attachment.
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.
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 to view 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 want to ...
Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.
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
Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!
I have the a2billing installed and Elastix MT, but cant make it works.
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 -...
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...
