We are using FusionPBX, in which, we want to play a IVR, which will be an announcement like: hello engineer, your site id: 12345 is down since 10:25 am, please reach to the site with address, 12 pavel street delhi. the severity is, 1, and priority of the issue is, 2. The numbers are coming from a data source every time announcement is played. this is done in fusionpbx/freeswitch using LUA code; as we have learnt. We need someone to write this code and give this functionality.
buenas, me gustaria un teninin completo de la plataforma astpp, la tengo ya instalada pero al configurarla me da error
buenas, e instalado astpp y le he echo todas las configuraciones, ahora tengo un problema y es cuando los clientes me mandan peticiones me cuelga el servidor, dejare la imagen anexa para que la vean
hi, we need below functionality, between BBB & FusionPBX. 1 SIP connectivity to Video Devices, based on IP(SIP based connectivity, based on IP. via pbx) 2 Dial in number connectivity for joining the VC, via fusionpbx.
we have our astpp platform and need configure
i need a multi-tenant system in fusion pbx and i want it to brand it with logo etc. a design change is required.
I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.
need a multi tenant system in fusion pbx. which can create extension for softphone etc. which can do all the things ip pbx systems are suppoed to do
I have an old astpp customizations and I need to update my customizations on the new server. I have the file with all the customizations and file edited, just need someone to re-apply on the new server.
goautodial using ASTPP trunk will having problem for autodial. customer answered the call but get silence and it wont pass to agent softphone. within 5 second will drop the call. tested using manual call with no problem with RTP voice complete. i has been try to use other supplier trunk and getting not issue with manual and auto dial from campaign.
I have running fusionpbx on cloud with public ip, need to connect Cisco phone as well as Yeastar gateway xml file is already uploaded but not registered
Install 2 ASTPP servers following this doc: ASTPP1 will handle billing, and ASTPP2 here will have a role of freeswitch only (handling calls). ASTPP2 needs to be connected to ASTPP1 DB, so that in ASTPP1 GUI we see also ASTPP2. Users from ASTPP1, must be able to call to user connected on ASTPP2. In ASTPP1 GUI, we must see the customers registered on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only); The done job must
Download free trial of this Modify this to work with FreeSwitch/FusionPBX and SuiteCRM. Will have a development server setup soon for you to use soon. Need it done in 4 weeks, place your bid and let me know how confident you can do this.
Estimados(as), Se requiere administrador de sistemas con experiencia en implementación de softswitch y sip trunks. MagnusBilling, ASTPP o OV500.
Dear All, I have a fresh installed FusionPBX on a Cloud [Ubuntu 20.04] and I have configured it to use TLS And the remote registered extensions haven't audio Note: the remote registered extensions are behind a NAT and are located in a country is restricted the SIP traffic so I should use TLS with SRTP to bypass the SIP blockage. Note: Award will be to the one have an experience only to complete this project successful (no wasting time) Thanks to all who will bid on this project!
Hello I am looking for someone who knows the programming of astpp freeswitch scripts. There is PHP script , lua script and XML script. Looking for someone who has already worked before with those files. I am not looking for web interface programmer. Looking for the freeswitch part programming
Hi guys, we run fusionPBX on AWS with an end user via grandstream ht801 ATA. She is US based and complaining about excessive Post Dial Delay (PDD) - meaning she dials the number she wants to connect to...then hears silence for 4, 5, 7, sometimes 10 seconds, then hears the ringback tone begin. We use BulkVS as our main provider for outgoing, however also use Alcazar and can switch to others. I have a VoIP phone on my desk connected to the same and am experiencing the same issue. we need to: 1. be able to measure the PDD 2. understand what is causing it to be so excessive 3. correct it - forcing a ringback to 'cover' the gap is acceptable if not other solution You can work with our SysOp, Harold, and he'll provide access and
Install 2 ASTPP servers following this doc: ASTPP1 will handle billing, and ASTPP2 here will have a role of freeswitch only (handling calls). ASTPP2 needs to be connected to ASTPP1 DB, so that in ASTPP1 GUI we see also ASTPP2. Users from ASTPP1, must be able to call to user connected on ASTPP2. In ASTPP1 GUI, we must see the customers registered on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only). The done job must be documented from the beginning, very detailed, how the ASTPP's were connected, after
Hi, I'd like to do the following on FusionPBX. Think would need to utilize the IVR then do a couple customizations? (1) Receive inbound call (2) Play recorded questions (3) Capture DTMF responses to the questions (4) Export out the results to a .csv (or store in a DB that can be accessed)
I need the phone number list in call broadcast to push through caller id name to agent dialer. Please note in docs above. The whole system works and we need it to push through the caller names as shown in link as example 1,2 or 3 to change it in caller ID Name. For example if the list was as follow 1234567890|John 3216549877|Mary 5198432163|Adam I want the John mary or adam to show when it pushes the call to the agents extension.
Hi Nasir I., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
Servers (CentOS and Ubuntu) will be provided for the right person. Connect freeswitch to astpp server, add a vendor and pass calls. You document all so that anyone with minimal tech knowledge can take the document, follow each step there and get the same results (freeswitch + astpp installed, freeswitch connected to astpp, calls route via a vendor). Then I will consider the project is done and it will be paid. Otherwise if we can not repeat the same from our side, the project is not considered as done and will not be paid. Please confirm, that all details of the project are clear we could go forward. Thanks.
We are looking customised multi-tenant open source call centre VOIP solution, with an android and IoS app for calling agents. We should be able to appoint resellers, white label it for resellers, we should be able to sell b2b call centre sol...management Graphical reports Multi-level IVRS Productivity Sale Graph Real-time call status Standard Call features Standard reports and other range of reports Voicemail All basic features Inbound number Black Listing Email & SMS Module Internal Chat Module Skill based Routing with Agent Ranking Sticky Agent You may use any or many open source softwares to get these features, like Asterisk Fusion Pbx, ASTPP, FreePbx, free switch telephony, webrtc, a2billing, vicidial, magnus billing etc. The system has to be robust scalable and capable to...
I need to develop an API to integrate Fusion PABX with my ERP, to validate whether customers are part of the base.
We have a project left halfway by a freelancer. Fusionpbx is already installed on server, just need to configure trunks, gateway with our SIP provider and create 10 extensions to be used on softphone.
Hi. I need ongoing help with FusionPBX. I prefer someone who is online frequently when issues arise like right now :)
I have setup a FusionPBX which is running on FreeSWITCH and I am using a branded Linphone as a softclient. However the iOS app cannot receive call while running on the background as it is expecting a push notification from the server. I need someone to setup the push notification on the server for iOS and android.
We just deployed a cloud based FusionPBX. We urgently need a freelancer to assist in completing the setup configurations. The underlisted tasks are what we urgently want. - Configure Gateway with a SIP provider. - Inbound and Outbound calls. - IVR setup.
I am looking for a front end developer to implement a frontend frameworks ie. Tailwind CSS, VueJS (or React) to make FusionPBX (open source project) have more of a clean and modern look and feel. Please send me a portfolio of your work that looks clean and modern, design as well as development experience required.
I have setup a FusionPBX which is running on FreeSWITCH and I am using a branded Linphone as a softclient. However the iOS app cannot receive call while running on the background as it is expecting a push notification from the server. I need someone to setup the push notification on the server for iOS and android.
The goal of the project is to install the free version of ASTPP on a VPS server. Then install a Freeswitch on second VPS server and connect it to ASTPP. So ASTPP server will be the slave+master+DB and Freeswitch will be the server handling the calls. ASTPP can be installed following this document: 2 empty VPS servers will be provided. All installation process (Freeswitch) and how it is connected to ASTPP server must be documented from scratch, step by step (how to connect to server, what to install, how, etc.), so that anyone with no tech knowledge can take the doc, follow it and install another Freeswitch, on an empty server, connect it to ASTPP server and make calls via real vendors. The project is considered fully done and
I currently have Fusionpbx running on a Vultr VPS. I need to migrate this over to my dedicated server on IONOS. I need someone to do this for me that really knows how to do it. This is not a time to train on my project. If you really know how to move it over let me know.
Warning: If you do not have any experience in freeswitch or Fusionpbx it is not worth sending proposals, like web companies. Only people / companies that can prove that they have experience Installation and configuration of WebRtc In the Back-End, I need you to create all the conditions for the front-end team to be able to create a contact center platform, such as: Create Agents Create Extensions Create Queues, Assign Agents to Queues Create outbound campaigns (power dial, manual, progressive, etc.) Create IVR Create domains Create recordings Music on Hold Mini CRM Breaks Reporting WallBoard Agent page, log in to queues and campaigns, answer calls with auto answer, mute, hold, transfer, script, request a break from the supervisor. I am available to detail the project.
Hello, We are hosting a FusionPBX instance in Vultr cloud. I would like to have someone versed that can help me get the system working properly in a test environment. I have outbound calling working but currently inbound is rejected, Most likely a firewall issue. I would also like to use the same person who helps me set this up for continued support on the system. I look forward to working with you.
Hi. I need ongoing help with FusionPBX. I prefer someone who is online frequently when issues arise like right now :)
I need SMS setup on my FreePBX & FusionPBX hosts, I also need documentation and training on the process.
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:
Hello i need a shell string for call from windows to FusionPBX Regards
The project is fully detailed at the following link, accessible publicly:
Hello, We are looking for a developer who is experienced in PHP rest API and basic knowledge about PostgreSQL. Basically, we are implementing fusionPBX with other CMS so we are creating some APIs we need to modification in existing APIs and create new APIs.
I need an expert who can troubleshoot the FusionPBX as outbound calls are not going internally and externally. Calls from any number ringing on the extensions successfully.
We are a communications company in south africa. We are looking at migrating our hosted services from asterisk to fusionpbx. We require assistance in configuration and require training so we can configure future deployments.
Hi NetworkLab, I noticed your profile and would like to offer you my project. We are a telecoms company in South Africa looking to migrate our hosted platform from asterisk to fusionpbx. I am just stuck in NAT-HELL and am in need of an expert in not only solving my NAT issues but also training me on future deployments
I need someone to secure my ASTPP billing solution based on FREESWITCH.
This task is straight forward. I need a iOS linphone app that recieve VoIP push and work in background. Only bid if you have done this before. No time waster. I will need to see evidence you have done this or we start with a proof of concept. Open to other iOS mobile sip client that can work with freeswitch and recieve push. I will provide server detail - i use fusionpbx
I need various work on several FusionPBX servers. Experts only who work efficiently. Ready to hire now.
We have a FusionPBX, Multo Tennant and the issue is when voicemails are left, they are not sending to the recipients' emails.
