Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDr...integrated into pipedrive are below. [login to view URL] Freepbx is an ASTERISK based phone system. Need to know how long this will take. Must have prior experience with Asterisk based phone systems.
...client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass
Hello, I need help in creating a process flow diagram on Aspen Hysys, or edraw or viseo. For the production of ethylene from syngas. As well as a material balance for the process.
I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.
System requirements Registration screen Logon screen online Users screen Server php and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple appli...
must be compatible with MS Access 2010. Need the database to be be able to track employees first and last name, hire date, termination dates, age, ethnicity, department, employee level or management level, tracking data for CPR training, TB shots, drivers license checks, background checks, drug screens, related costs of new hire training, and duty position
...[login to view URL] Actually I am developing a web application using AdminLTE. Now I have a scenario where I need two additional requirements. One is Form Builder and another one is Process Builder. Please find the screenshots in zip folders under project description. In both the cases I need two features. When I press save button then whole design should come
small task build program using Parallel process skills .
I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.
Design a Raspberry Pi compliant HA...Accelerometer MCP3808 ==> Analog Digital converter MCP2515 ==> CAN bus controller MCP2562 ==> CAN Driver 16 Mhz Oscillator for the CAN bus 120 Ohm Termination Resistor for the CAN Bus termination selectable via Jumper 10 PCB Screw terminals for the 8 Analog Channels of the MCP3808 and 2 for CAN High CAN Low
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be completed in order for push notific...
Key Duties and Responsibilities - Interview the key process owners and members contributing to key processes - Obtain knowledge of about existing processes and document in written form - Incorporate process flows with key control points - Reviews and discuss the policies with key process owners Requirement - Based in Singapore - Diploma /
Script needed to read file name from file and rename it: Input: read text file with name of the file in format : cam1_hd_(16-10-2018_19-55-00).mpg Need to rename this file to this format: [login to view URL] Date and time format is always the same, but prefix can be different: cam2, cam5_hd, cam_23_low_res, etc, and underscore from prefix needs to be removed. Platform: CentOS 6, BASH.
i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test
We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux
...modify those. Below are more detailed descriptions of the api/ and frontend/ directories. Locations that you will be mainly focussing on as you develop are marked with an asterisk (*). Backend API (api/) api ├── Dockerfile # Description of Docker image ├── [login to view URL] # NPM dependencies ├── spec # The backend tests
My colleague and I are writing a poster for presentation at a pharmacist conference. We require two professional-looking graphics of a seven step process as a cycle with a person (patient) in the centre. See images attached (created in powerpoint) - we'd like the graphics created using graphic design program (i.e. must be better than we can do ourselves
Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined
...currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from a paid PBX (switchvox)
To resolve issues in an existing Magento 2 website that's in development. To work on further web development related tasks. Preferably speaks Russian.
Hi, i am actually working on to identify the roles of the architect, engineer and quantity surveyor in a building project. I am needed to speak to a practicing c&s engineer, to know about the real life relationship between the 3 professions. You will be needed to explain about your conflict/ job overlapping with the over 2 profession, starting from the preliminary stage > design stage > ...
i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]
...english speaking countries. Need someone experience who can guide us with all basic info needed to start the process. We are ready to pay one time setup fees for each process he/she will setup for us.. 1. Educating us on how to run the process. 2. from where we get data. 3. how we will convert data to an paid lead.. 4. how we can sell lead and get paid
We have SaaS solution for which we need to switch from Chargebee to Stripe and create a ...solution for which we need to switch from Chargebee to Stripe and create a payment page. In the meantime, we will also streamline the on-boarding process according to the attached wireframes. This on-boarding process also includes sending datalayer to Intercom.
We are having a requirement of web application which will involve following modules/process: CRM Recurring alerts and schedules to both customer and company for service and payments Quotations generation involving trivial calculations and robust market rates
Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem
Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward
...forms and documents on our Web Site. The documents are made with LibreOffice and changed to PDF for use. The server used has 3 drives. One for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must
I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.
Need technically qualified and experienced technical content writers to write about BPM - Business Process Management and BPMN. They must have past experience in any BPM software or BPMN process modelling. We need around 10 topics on BPM to be written. Each topic will be around 500-600 words of original and engaging content.
**Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development
I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?
...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message
We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information