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 DevelopersLooking for android VOIP dialer with anti blocking solution, need to work of blocking countries. audio and video it's should support all kind of soft switches
I have a raspberry pi with a working asterisk 16 and chan_dongle I am looking for someone who can 1 capture dongle data from ami and update in db 2 assign static names to each port on my usb hub so that port 2 will always be port 2 3 insert dongle imei in a mysql table based on which ports the dongle is in 4. call a file if there has been an imei change on a usb port
Hello Everyone i need My server New User Singup Option my website Can Help Me Just Contuct Us ..... My website : [login to view URL]
plan and design a highly scaleable, fault tolerant IT infrastructure for my company on amazon web services. - Hosting enviroment for my websites -VOIP currently use Ooma office - Google resources for virtual office enviroment - Email server - Streaming enviroment
Need an expert in VirtualBox and Androide-x86 installation. The job will using TeamViewer.
I want a Power Phone Dialer built similar to Phone Burner. I am a CRM developer and I want a product I can resell to my clients and have it integrate with Bitrix24 with a Rest API. Do not contact me if you do not have a telephony experience. You are wasting my time and yours. Thank you!
We're a VOIP phone company. We need to setup an API to talk to our carriers that when a number is ported out we get an email say XYZ number is porting out do you approve yes or no. The Port-out Validation API allows approved and contracted customers to confirm the validity of port-out requests on submission, to ensure that the requests include the required FCC-permitted elements of subscrib...
Hello i need voip billing switch & anti block mobile dialer android + iphone with whats app [login to view URL] you have ready with testing complete sent details demo link & price details. my need reefer option in dialer reefer account holder found profit % from admin & profit balance share option another user profit account .agent route buy sale possible freedom. Another requirements ...
Hi freelancers, I need one voip expert for urgent basis. YOu should have clear knowledge how to set up voip service for call center. You should have clear idea bout DID, SIP TRUNK and other terms. I want to set up 3CX voip services for my company. YOu have to add toll free numbers to 3cx so staffs can pick up calls from their apps. Thanks again.
I am using Elastix 2.5 (CentOS 5.11) with 4 NIC's (1 for WAN and 3 For SIP Trunk using IP Authentication) eth0 - WAN IPADDR=75.XX.XX.162 NETMASK=[login to view URL] GATEWAY=75.XX.XX.161 eth1 - SIP TRUNK1 IPADDR=100.XX.XX.198 NETMASK=[login to view URL] GATEWAY=100.XX.XX.197 eth2 - SIP TRUNK2 IPADDR=100.XX.XX.194 NETMASK=[login to view URL] GATEWAY=100.XX.XX.193 eth3 - SIP TRUNK3 IPADDR=10...
Works on Chrome, Firefox, IE, Safari, Opera and Bowser Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat) 3GPP IMS standards
SIP Voip APP example Bria Platforms Android Apple MAC and Windows
I would like to integrate Exotel system (based on OpenVBX) into my FreePBX server for seamless inbound and outbound calls. Freelancer must have experience to integrate the APIs from httpsdeveloper[dot]exotel[dot]com/api/ as a Trunk and utilize other applets from httpsdeveloper[dot]exotel[dot]com/applet for managing calls to ensure the combined system effectively helps to stay in touch with our...
CloudSetup Installation Configuring Minutes Current Setup should be for 10 agents 3 Campaigns - 1) Press 1 IVR set Up for Inbound Campaign 2) Outbound Telecalling 3) IVR Broadcasting Budget would be low as 5k
I want a new asterisk server built to replace our old version. i want someone who will build the server from scratch. Not just copy the sample files and fiddle with them. i require the following functionality. 1. Twenty Extensions. SIP Phone and Softphones etc 2. Conference bridge with IVR to Choose a room. Must support Video and Messages. 3. Short text message between extensions. 4. Video Cal...
We need an application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. General Deliver...
I want a power dialer similar to Phone Burner. I want to be able to integrate it into my CRM with an Rest API integration tool.
I want a power dialer similar to Phone Burner. I want to be able to integrate it into my CRM with an Rest API integration tool.
Hello, the script already exits. There is a documentation. I need a full integration of the script in Node.js, one side FreePBX and the other Amazon Lex (or google speech). Possible upgrade with UniMRCP. here is the link [login to view URL] here is the explanation on how to configure [login to view URL]
i need do limit of the call on asterisk like 16 calls max at same time and i need update this limit easy when i need by asterisk database
We are using SIP/PBX setup (Kamailio-Freeswitch) with Kazoo. We need to create a User page where 2 or more people can have video conference with least settings. Currently Phone and Fax is working. I think video work on certain apps.
We are looking to develop a replica of [login to view URL] urgently. We dont need all the features from Zoiper. We just need a mobile application (IOS and Android) that can make a phone call directly to a land line or mobile number. Experience in voip applications/ softphones is required.
we need to prepare a linux OS that will include Armbian with asterisk 13 and vpn to be installed on a RK322X devices you can find here the link how to do it [login to view URL] our budget is 100 dollar
Multi-tenant voice broadcasting system. Multi-tenant press 1 campaign. CRD report with billing details. (WHMCS Billing Module) Integration with Sign up page, So whenever new client get signup they can receive email with voice broadcasting credentials. If Possible. Or need Custom On pressing 1 or pressing Any call should connect to call center agents. DID management based on the campaign. Fe...
Configure latest Vicidial 1 Indian VOIP line where when user logs in they get call on the cellular phone to bridge in conference. {Milestone 1 1000 Rs } 1 UK Line to dial out UK Number {Milestone 2 1000 Rs} 1 Additional UK Line to dial out UK Numbers where fist line do not work. 1 USA line to Dial out USA numbers Configuration should be such that dial er should automatically pickup lines ba...
I need a software solution that allow my employee to attend calls from home. This includes IVR, Customised CRM , & Call recordings. Option to make outbound calls is needed along with bulk file upload from the admin panel.
I am looking for a parallel code to be written to the phone systems prisons use for inmates to call out. They are usually prepaid phone systems. An example is GTL. Can we use the phone system and charge a cheaper rate?
hello I hope you all are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. Budget is tight as well as time schedule but there is a great opportunity also for the maintanance phase.
Hi We have ASTPP billing installed and configured as a gateway (resource) for our VoIP system, currently, ASTTP uses IP Authentication when we define a client, this result in an issue that all traffic comes from our FreeSWITCH pointing to this ASTPP (gateway) is considered as 1 client based on that IP and the balance is deducted from one customer no matter the user who is making the call. We wan...
Bid if you are familiar with softphone and web apps
Theory of operations: 1. Customer will dial a satellite nr 2. Asterisk will recognise the pattern and will add an additional string in front of the number
The SIP DID/Call Forwarder System is a simple system that will take an inbound call to the system and depending on the DID/Destination being called (based on a SQL = MariaDB or MySQL) will forward the call back out to another destination. The idea is that this sits on top of asterisk. This should be 100% database driven. The system will also need to be self-installed, so we can run ideally 1 scri...
Will give more info in chat
