Vicidial freeswitch jobs
Need a custom CDR and DID report page. Please DM me for more info. Please note that a video interview is required before this project begins.
I want to learn how to setup vicidial. Like creating campaigns, Creating an Outbound Campaign, Creating a Press 1 Campaign, and so on.
- Develop CTI pop-up in salesforce and upload it on salesforce marketplace - Use or any other method to manage SIP registration, Call, Recording and other necessary features(Outbound call, Inbound Call, Blind Transfer, Attended Transfer, Mute, Unmute). - VoIP PBX is based on FreeSWITCH Language -Javascript, Node.js, APAX (Or any other needed for salesforce CTI pop-up)
Hi Ambiorix R., I am looking for a very simple custom report for my new Vicidial server. Basically I want to combine the agent time detail and agent performance detail reports and add two mathematical fields which would be net time (total time minutes pause) and calls per net hour (total calls /net hr) I am willing to pay extra to have it turned around 24-48 hours. I have a lot more work coming down the pipeline if all goes well!
Hi Milos - I just moved my call center to a Vicidial call server and I would like a custom report built. It should be pretty simple because I want two of the stock reports combined in to one with only 2 new fields added, both fields are functions based off of data already pulled for the report i.e. net time = total time logged in minus pause time and calls per hour = calls divided by net time. I would be willing to pay extra if you could get it to me by 4pm PST 02/08/23. Thanks!
We are looking for freelance expert who can work an a specific set of agenda to explain and deliver it's outcomes
Hi, We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client. Our flow of calls is like this: WebRTC client -> OpenSIPS -> FreeSWITCH The system is deployed on Azure. We are looking for experienced person who has done such work and quickly help us.
Setup Vicidial Full configure with Telnyx or Pilvo VICIDIAL full setup with SIP trunk - we provide server.
Hello Freelancers, I’m looking to implant live DTMF logging into my agents panel using asterisk. The DTMF will be read from asterisk server and will be shown in my agents screen in real time.
Hello all, I would like to capture live DTMF information through asterisk and make it appear on my agent login panel. Essentially all it will do is when a call gets connect and an agent ask the customer to enter a code through their key pad it should appear on the agents panel.
What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time (9a.m. ...
Freeswitch amoCRM intgration to enable voice call on the amoCRM
Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.
We would like to setup Signalwire video conferencing and integrate it with our HoduPBX freeswitch multi tenant platform
...a step-by-step guide for configuring FusionPBX/FreeSwitch to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown which will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN ()
Configurate carrier, manual and predictive dialer for France
We need a so-called "banner information" for our own web-based control panel, which is connected to the FusionPBX / Freeswitch telephone system. This means that you can double click on the visually displayed subscribers and then add text there, which is then visible in the banner. In addition, background color and font color should be customizable. Also a link is to be opened by means of right-click and "open link" or so. The text, which one can write into the banner remains to be seen so long on the surface, until the call is separated. These data like URL, color for the background, font color, etc. should be extractable from the description field from the destination management at Fusionpanel. That means that we enter there a hexadecimal code color for the bac...
This project is about customising Vicidial & Asterisk applications. + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
I am looking for a lua developer who can help me to customise something in my VoIP server. It's lua script which do the functions.. and also use mysql database. Experience in the freeswitch server will be an advantage. Thanks
Hello, I need to make a skin for vicidial for press one calls, I want a comfortable environment, I see that you have several panels designed and I want to know the costs. IVR Press System Request - 1 1- Create users (Clients) , When creating a user you must create 15 extensions that will only receive calls from the press one that has that user enabled, When the user enters his panel, he will be able to choose the bells pre-assigned to his profile... Upload the listings to their profile and they can manage it, They can see their extensions online, in call, or disconnected. You can check the available balance with the provider (it will be my voipBilling). The system will have 2 types of listings, one for customers and one for caller IDs. =========================================...
We are looking for an engineer who can assist with FREEswitch Fusion PBX. We need to update voice messages for 5x PBX users (Christmas greeting messages). Change extension labels for 5 customers and install new extensions and hardware for 2 PBX users. If you can offer other services around FREEswitch Fusion, we would be interested to learn your skills and what you can offer to us.
This project is about customising Vicidial & Asterisk applications. + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
Vicidial Goautodial configuration I need to setup carrier for inbound outbound callings
Hello, we are a hosted voip service provider and until now we have been using FreePBX and Asterisk, we are looking at moving away from FreePBX. We are open to either Asterisk or FreeSwitch, whichever meets our requirements listed below: Good day, We are a VoIP Solutions Provider that is currently looking to move away from our FreePBX systems we host for clients to a Class 5 Softswitch/PBX. I have compiled a list of features we want but don’t currently have with FreePBX and then the top features of FreePBX that are an absolute must have in this development. This would need to be a linux based platform capable on running on multiple dedicated servers with iSCSI storage. Ideally, we would like the platform built on a AlmaLinux or Rocky Linux. We would want the ability to ha...
Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.
WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk
Hey i have installed Vicidial on Hetzner but am having some trouble first am getting netowrk is unreachable, do you think you can help me with that task.
Hi, We need someone who can upgrade our FreeSWITCH and OpenSIPs to the newest stable versions on Amazon AWS. Currently we use FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is 1.10.8 We also need OpenSIPs upgraded to the newest version 3.3.2 we currently are on: 3.0.2 (x86_64/linux) This is a live production server so it will need to be done pretty quick in a couple hours or so. If we work well together I will have many more ongoing tasks involving FreeSWITCH, OpenSIPs, our PBX and other issues, our main telecom engineer/developer was in Ukraine and we have not heard back form him in months. Thank you! Thank you!
+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
Hi openasterisk, I noticed your profile and would like to offer you my project. We can discuss any details over chat. https://www.freelancer.com/u/openasterisk/portfolio/vicidial-custom-themes-for-agent-and-admin-portals-2013776
Hi, i am looking for someone that can assist me with some small setups on vicidial. I have already setup a vicibox solution on the server and have a sip trunk. We did use fusionpbx for a small call center with only outbound calls but looking to move to vicidial. I am struggling to setup the carrier and have issues with the time sync on the vicidial agent side. Our time is out with 10min for some odd reason. I have already checked the files in apache and cli which is correct. the settings in vicidial is also set to +2 for time zone. I am looking for someone that can assist just to setup the sip trunk and also make sure that the agents can dial out with the webphone that is built into vicidial for now. Will you be able to assist today and what will the cos...
We are looking for VICIDIAL Call Center-CRM Set up And Training personally in Hyderabad. Inbound and Outbound all scenarios with WebRTC agent, need demo and CRM integration
Hi Arshad N., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
Hi Aqs Y., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
...using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. Our Soft-phones are made with React.js I need a person who knows what they are doing, we also use OpenSIPs so the codecs in OpenSIPs might not be correct but this is just a guess. Can someone solve this for me, I have a hard time getting honest developers here, it seems like everyone says they can fix it, then I waste a week with them and have to cancel and look for a new developers, please only bid if you are truly an ...
Various tasks in freeswitch. Requirements - understanding, communication, desire to progress on the subject. long-term cooperation
I need to install vici dial + additional CRM for inbound / outbound calling. Except the physical server I need it to be in the cloud for accessing the GUI. Also for outbound calls I have goip with 4 ports.
+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
Develop a new pbx system voip with modern interface (need to have images to see) where we use didww for the numbers which are ready and setup call routing, music on hold, setup call conference and schedule and rest is all normal routing and dialing. Call flow is available. Freepbx or vicidial or any other system that is modern and easy to setup as it has to be ready end of the week.
We want to provide our cloud switchboard solution via freeswitch. For this, I would like to discuss the project with people who have experience in Freeswitch and API development.
We currently have an asterix pbx without any userinterface. We want to have vicidial added with a fresh new custom theme that you have to provide. The call flow is in 3 languages and we have 7 websites to connect but that is all in the asterix pbx already configured on a dedicated server.
Connect the Web Application with API and Dynamic Data with the Freeswitch / FusionPBX System.
+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required
Vtiger integration with my Vicidial All emails, notes, recordings and tasks to go inside each lead. Admin needs to see recordings inside each lead and agents do not see recordings. Branding- Remove Vtiger branding Vtiger integration single signon Callbacks should be integrated with tasks in vtiger. Outbound email is not working on vtiger so needs to work when you set it up. Inbound email is working fine. 20 Custom fields that need to be on agent screen and importable and exportable. Full walkthrough how to use it Budget is $50 and will lead to a lot of installs for my clients and maintenance work for Vtiger.
Hi I need a hand with freeswitch: I have a client that needs to send calls to my freeswitch but his switch doesn't have username+password authentication. He is asking me to have my freeswitch accept the calls based on his public ip address. Let me know if you are able to help but only bid if you have extensive experience with freeswitch as this is a security concern. Max 100 euros. Thank you.
...Description • Candidate should be familiar and comfortable with Freeswitch. • SIP Development experience. • Must be aware of Sip and webrtc integration. • VOIP software development. • Good Knowledge in PBX, SIP, RTP protocols. • Worked on Queue, IVR and Voicemail related applications. • Expert in Freeswitch installation, configuration and... • Competent enough to setup daily call limit and concurrent calls Requirements · Software Development experience in Freeswitch, FusionPBX, Opensips, SIP, VOIP, SDP, TDM, IMS, PSTN, Python, Perl, Linux, and Open Source Technologies. · Strong Technical, Logical and Debugging skills with innovative and result-oriented approach ·working experience in Python, Shell, Pe...