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 Developerscall center software complete for incomming and outgoing calls. AI agent-based system that can place maximum calls, based on script and provided answer from prospect wil proceed a option from script. when al ok will redirect to live agent. when no intrest wil hang up. prefer open source software. AI solution is for Europe.
To make interchangeable Invoice and Credit Templates. The Templates are made by us with LibreOffice and must be changed to pdf without loosing calculations. They will be filled out on our server, to be send by email. *** Requirements: 1) LO Calc spreadsheets for addresses, with interchangeable background image of Invoices and Credit Notes. 2) Software to change the ods files to pdf, without loos...
Hello / Good evening everyone, I would like you to help me create an asterisk call file which places several dtmf commands in the call file channel once the call is connected. The DTMF does not necessarily have to go through the call file. If you want you can use the [login to view URL] file, but my goal is to place dtmf commands in an ongoing call generated by a call file.
Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile ver...
I need to develop a platform to send bulk messages through web interface.
Full Asterisk IVR Setup training and cdr report with below requirement 1. Creation of IVR flow chart and diagram 2. Communicating with other stakeholders for the Audio files. 3. IVR Configuration on FreePBX 4. Setting up Inbound routes to point to IVR (via GUI and Command line) 5. Other tweaks on IVR (schedules / time conditions) (via GUI and Command line) 6. Setup of dial plan on GUI and CLI 7. ...
We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of telephony systems based on these platforms -Excellent understanding of pbx telephony, dial-plans,...
Issues: Recording not found. Recordins are not process and published. Link of recording cannot see in chamilo. I need fix problems and records session. BBB is working right now.
If you can create an add-in in outlook and understand SIP/VOIP please reach out.
Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. Thank you.
To make many Invoice Templates for inbound DID billing. Open source software is to be used for this project. The Templates in pdf are to be filled out on our server, to make the invoices to be send by email. Required: 1) LO Calc spreadsheet with interchangeable background image of Invoices. 2) Software to change ods to pdf without loosing calculations. 3) Asterisk CEL to be used to fill Cells...
Hi, I need to set up a rule that in FreePBX that will allow changing the CID of the extension based on the outbound route. For example, if I dial out from exten 200, through outbound route 1 it will get CID1, outbound route 2 - CID2, etc. The outbound routes are NOT defining the CID, the extension defines the CID.
Device Compatibility: The App will be compatible for Android devices. Scope of work : ● revamp the open source code and redo the entire application ● Remove/ disable extra functions ● Revamp the design ● Update new color combinations and logos ● Test the application ● Deploying It to store SDK linphone
I am in need of resources and tutorial on kamailio configuration file for serial forking with tutorial
I have installed Astpp on centos 7.x server and need to confiugre it with fusionpbx that is installed on separate server , you need to perform following make setup functional and test create user / tenant / domain add sip trunks create extensions successful testing cooperation rebranding etc
Hello! I'd like to invite you to take a look at the job I've posted. Please submit a proposal if you're available and interested. Telecom specialist We are looking for system integrator who can help us to integrate facebook magma open source EPC with Mikroitk Enodeb .Its 4G LTE Private network project. We have one test unit from baicell which is recomanded by facebook . Our ...
i need to stop fake ringing on linphone application i used sip account to make a call
I am having problems with extension loosing register, i dont have a very good internet connection but i am almost certian that i can fix this problem, customer has 50mbps upload and download, the problem is the pings not being very stable, due to a wireless ptp connection, freepbx / asterisk setup.
I have sms working with flowroute and my fusionpbx. But right now its only working in one domain. I would like to make it work will all my domains and each domain with its own flowroute account. I have test server prepared for the task already.
Hi, We have a PRI line and connected to SIP server using FreePBX. Currently our employees working from home and using some soft phones but there is no voice clarity. We would like to setup a virtual callcenter which redirect to their mobile numbers. If this is possible please reach me.
Need someone available in KSA Riyadh, We need to configure the Yeaster Part .
We have a project to integrate odoo with 3cx so we can make voice calls from the webpage (WebRTC) and also when receiving voice calls to get the contact data on the screen
Looking for business pbx phone system. Cloud based here is a reference website [login to view URL] Mobile App Desktop App call waiting conference calling call transfer auto attendent extensions etc
Hello! Looking for someone to write an app to auto dial into a conference call. I can give more details once we move forward. And dont ask what language I want it programmed in, If I knew that I would do it myself :) You tell me!
Hello, I need someone to help us by setting up an inbound call center using Bitrix24 dialer. We have our own sip voip trunk and can also host freepbx vps if needed. We need to achieve the following call flow: Customer calls —> pbx places call on hold while waiting for InGroup member to be available —> once InGroup member is available call is routed to them in the Bitrix24 diale...
We have a lot of gateways for SMS services. All of them acept AT commands but, only a few have a VOIP access. (SIP protocol). For that, we need an app to convert SIP protocol to AT commands for termination of VOIP calls over SIMs through AT commands.
hello i am looking for someone with freeswitch knowledge person with astpp (billing) will be very good. i need something to do with SIP TLS with letsencrypt and other voip work.. so you need be very good on freeswitch for this
Backend system of IVR needs to be built This would include creating and integrating APIs with the IVR and Database (which also needs to be setup)
Looking for Asterisk expert who can help configure PBX. This task probably needs LESS than an hour work in this case my budget also mirroring. You need to able to talk.
We need a solution to replace our FreePBX setup. We currently have 4 instance of FreePBX, but this is not the perfect solution. Instead we would like two Class 4 softswitch for redundancy. in the front to give LCR and SBC fonction. We figured that Kamailio is one of or best solution and would then add Class5 switch, We would think that FusionPBX is a better solution than FreePBX. The setup need to...
Need to install sems-server on Debian server side and Openwrt client side. UAsL <---SIP/RTP---> SEMS-SBC1 <===SIP/MUX RTP===> SEMS-SBC2 <---> UAsR SEMS-SBC2 will be behind NAT. [login to view URL] [login to view URL]
as title need asterisk guru support for a call center project and little small distribution
Hello, Looking for SIP -> Whatsapp call gateway, it should take the calls from SIP (Freeswitch/Asterisk) and terminate to whatsapp number. Status of number should also monitored on whatsapp that means, if number is active on whatsapp, furthermore if it's getting the rings or not. It should also return the correct call error codes CALL SUCCESS, BUSY, UNAVAILABLE, etc.
Make calls with a list of numbers on my android phone, when people gets the phone play an audio and receive if people press for ex, 1, 2 or 3 on their phone. Call is made with a standard line. Not VoIP
I am a system integrator and manage systems for my customers . I am setting up a VOIP system for one of my customers using freePBX and would like to get assistance in setting it up. I am familiar with freePBX and how to use it, I need someone who can assist me with troubleshooting and provide additional support when required. Ideally this would suite someone already in the VOIP area and currentl...
Hacer modificaciones a la plataforma de Issabel: 1.- Poder identificar quien corta la llamada, cliente o agente 2.- Marcar hasta 4 números por registro 3.- Mensaje emergente cuando el agente recibe una llamada de una cola especifica 4.- Hasta que el agente guarde datos le caiga la siguiente llamada 5.- Bajar el tiempo de revisión de llamadas agendadas 6.- Poder tener hasta 3 llamadas...
O sistema do Issabel ele lê do o nome do cliente se tiver em uma lista? Ele ouve a resposta do cliente como sim ou não e me computa isso?? Preciso que leia o nome do cliente e depois fale o valor que o cliente tem disponível e depois ouça a resposta sim ou não é me dê um relatório. Tenho uma empresa de empréstimo consignado
