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.Palkkaa Asterisk PBX Developers
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.
Hello, Looking to get a previously working script modified/updated to work with new underling change. The script was written to work with Asterisk 11 and we are not running asterisk 13 and due a change in code the script no longer works. Phones are SPA5xx The scripts checks for calls that are parted in a specific parking lot and then displays all the related parked calls. the script is attached ...
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.
Looking for business pbx phone system. Cloud based here is a reference website [kirjaudu nähdäksesi URL:n] 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
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...
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. [kirjaudu nähdäksesi URL:n] [kirjaudu nähdäksesi URL:n]
as title need asterisk guru support for a call center project and little small distribution
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
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...
Hi, I need an idea or a tool or a configuration or a new codecs for VoIP Switch so as to the caller when hang up the call so the call don't hang up and continue online, connecting and counting on both side up to 3 minutes or more unless the caller switch off his mobile. It's like catch the call. For more details or question, so please contact me. My sk.ype... [kirjaudu nähdäks...
I need Topex MultiAccess GSM Gateway configuration. Following steps are requied: - Setup SIP Trunk in Topex MultiAccess GSM Gateway and IPX Huawei eSpaceU1960 - Configure outbound and inbound routing in Topex MultiAccess GSM Gateway and IPX Huawei eSpaceU1960 - Testing - Document configuration - Speak Spanish (prefer)
I'd like to complete pending setup of extensions, ring groups, dial plans, other modules that are needed for a school's internal / external communication. Require experienced VoIP admin with hands-on usage of FreePBX
I am going to start a project based on astpp and fusionpbx. I am looking for someone who have great knowledge on those two opensource platform. I will pay per hour and person mostly will be help me on chat. and also remote login to my pc if needed. I know 75% how to use them but before I go for business I need someone beside me. hope you understand my plan so if you think you are the right person ...
I need a program that I can make a outbound call by calling my Twiliow number from my landline and then entering the number I'd like to call after being prompted. I also need my landline number to show up as the caller ID not my twillio number
Make software, to be used with open source Billing software (to be discussed), using custom made invoice Templates with fillable fields in pdf, to be filled out on our server. The Templates are made by us with LibreOffice. Server and PBX (Asterisk) are on different machines and networks, with different IP addresses ! All is Linux - no MS ! You must be very familiar with open source Billing softwar...
I want to install Goautodial v4 in my VPS (Digital Ocean or Vultr) to have it functional for automatic calls, I also require the lists of leads, not to be loaded by the GUI, but to be loaded automatically from my LEADS manager, that my clients when leaving their numbers on my website, the calling system automatically calls them and links them to my operators.
Quiero instalar Goautodial v4 en mi VPS (Digital Ocean or Vultr) tenerlo funcional para llamadas automaticas, tambien requiero las listas de leads, no sean cargadas por el GUI, sino que se carguen automaticamente desde mi gestor de LEADS, que mis clientes al dejar sus numeros en mi sitio web, el sistema de llamada los llame automaticamente y los enlace con mis operadores.