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.Rekrut Asterisk PBX Developers
Need someone with 3cx expertise who can setup a contact centre call flow for me for testing. Covering ACD, Agents, Supervisor, Queuing & Call Reporting. Need it for 3 agent only as this is proof of concept.
- Setup a dialler on my server - further setup the dialler's campaign - dialler with recording is preferred (3rd part app with recording is also welcomed) - find a VoIP sip for this dialler's campaign with an unlimited budget plan for both inbound and outbound calling
We basically need a regular vendor for VOIP Carriers on regular basis.
I need a systems technician to set up a linux server with firewall, email, VoIP and Messaging services. The server will be either a phial server or a virtual server with a recognized service provider
my website is developed using CMS caller Perfex and I need you to integrate softphone so that my team can call leads using CRM only. Softphone Type: -with SIP login - work with VoIP channel with call recording
I want to setup ViciBox on a cloud server. I am thinking of using Google Cloud Computing, but if there are better alternatives that are more simple and more cost effective, please let me know. Also, I would like to link my domain name to the server where I can login from my domain name with a username and password. Please let me know how much and how soon.
Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. Please let me know only if you know how to manipulate "SIP Response Code" in Asterisk SIP Trunk. Thank you.
I'm looking for a service to install Asterisk or Elastic PABX for my call center Must have recording feature and reporting module, also looking to integrate this PABX to existing in-house CRM
Looking for someone who can reset password and configure Alcatel Call devices and Patton device. Please find this equipments in the uploaded files section, also note that awarding of the this job will depend on if you know the device it modules and functionality of each module, we only want those who have had experience working with this devices thanks.
We are having some issue with OLD Analogue Phones. The call is answering right after the dial is initiated. So our customers are being billed for hearing the ringing tone. The caller gets the 200-OK message at the beginning of the call. None of above is expected. We need our customers hear fake Ring back, not billed & their CDR not updated as answered until the call is picked up by a real hum...
Dear Candidates, We are looking for an asterisk, Freepbx and AWS cloud engineer who can take the role of supporting a telephone firm in a particular time. The engineer will require to take support during the +GMT 8 time zone. With that timing, the company allocates specific service level agreement to customers. The infrastructure consist of extension registered to assigned FREEPBX machine (EC2 i...
Hello I have issue with my Asterisk certified/13.18-cert2. it 's rejecting the incoming call with the header Tel: ... I need a small modification on the asterisk to accept the calls. I need developer who will develop small patch for me . find the attached invite
IVR Industries (and sub-industries) Manage audio lists by industry 1. Menus - each menu has playback file to navigate the listener what button to press. 2. Menus - ability to make sub menus. 3-5 levels. 3. Menu can navigate to: a. Sub menu b. Playlist of files located in the server c. Connect to Icecast Live Broadcast url to listen to radio. 4. Ability to insert "PROMO" files before ...
Hi. I'm running a SIP Trunk with asterisk. The problem is when we call a Analogue Phone the duration starts at the beginning of the ringing. We need someone who could setup the Ring Detection module of asterisk to bridge the call and let the duration start only when the ringing stops. More details will be described with the selected freelancer.
Hi, We are in process of building app on Android. When making audio video call users can't hear incoming or outgoing calls. User can't hear other user as well. We are using Twilio Api It was working fine but now app is on play store we are getting this problem. Need urgent fix only who is experienced. No time wasters
Cold calls, Need a predictive dialer. dial 100s of contractors and ask if they do work in our area and then just say "ok ill have the boss call you in 2 min". a lot of the phones dont get answered so we need to try them repeatedly through the day at different hours
kannel expert don't waste my time , I have two apis one to send sms ( we need to save variable name mms) second we need top dlr-url=%mms that's it I need someone who knows how to parse
The goal of this project is to modify/adjust fields under Customer Profile available in ASTPP billing according to our needs for import of customers in correct format: 1. Enable possibility to have special characters and Russian letters in required fields when importing customer list. All details of the project are available under this public link: [login untuk melihat URL]
We need someone who can find our call centre new projects to work on, in order to be paid you will need to provide me with a contact for a new project, once the deal has been done and we get our first payment you will be paid.
I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tena...
i need integration erpnext crm with issabel and make pop up on call show customer details and call record and ather and i need sorce code We should have incoming call popup with data being picked up from the ERP Next CRM Module for a existing number and We should have a option to save a un-known new number as a new Lead or Save to a existing customer / supplier / other contact similarly we should...
Need a tech or developer to tune up Fusion PBX [login untuk melihat URL] some registration issues on Legacy devices like Cisco [login untuk melihat URL] issues with ATAs, Cisco SPA 112 & Grandstream ht801. Also need to whitelabel fully.
Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.
Please let me know if you have experience in above tools. I've some little things to do. First project is. I need to mask the RTP IP of the originating site with any IP which is not present in the server. So the termination site will see a different RTP IP than the original one. More details will be discussed on chat.