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.De 19416 comentarios, los clientes califican nuestro Asterisk PBX Developers 4.81 de un total de 5 estrellas.
Dear all, as per title, I would need to get FreePBX installed on a cloud host, setup and configured with 1 landline, ability to send/receive text messages, voicemail and IVR. As a softphone I would then use Zoiper running on Linux Ubuntu. I would need your advice on choosing a reliable cloud hosting service, sip trunks and landline numbers provider. I would also require an initial induction on managing FreePBX on my own, how to add additional numbers and edit the IVR when needed. Many thanks for your interest, I look forward to receiving your proposal. Al MAINPLAY
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.
I am using FreePbx server and - (1) I want to make robo calls to phone numbers automatically and deliver a voice message. (2) if somebody picks the phone and responds then a lead has to be created in our crm system. (3) I want the ability to limit the number of calls which can be done simultaneously. (Like 3 or 4 calls concurrently)
Need experience with VoIP SMS please send me examples of sites and interfaces you already have All the requirements can be done with open source like But with a change of the front part and also need to add some functions there as well
I have a white-labeled Linphone application for both iOS and Android completed and use Asterisk servers. Push notification for incoming calls is causing me challenges. I'm looking for someone to hire to: Walk me through setting up my Apple developer account and Firebase to send push notifications to my Linphone build. Walk me through associated modifications to Linphone build (integrate Google plist, etc, whatever needs doing). Configure kamailio (preferred) or flexisip (acceptable) to proxy between my Asterisk systems and Linphone on client mobile devices to handle push. We'll use a fresh Debian 11 install that I will provide you credentials for. If this is something that you can take on, please provide a quotation.
looking for developer who knwos about any software that it can be installed on my PC to install it for me the software must use as respeach instantly voice to voice using some body else voice which i can upload in to the software to replicate, but must send me sample of the voice and the coloned result first to check the accuracy of the software
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 versions of the application, no matter the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, that triggers successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to the WhatsApp gateway 2) WhatsApp gateway converts the SIP/...
1. Vendor will provide the onsite implementation services (Hyderabad Location) of all in-scope project hardware and software (Attached BOQ). Services to be provided are summarized in the following sections below. 2. Provide the complete planning, & Implementation services for shared BOQ. 3. Implementation of this SOW in accordance with customer document 4. Installation of Virtual CUCM (Unified Call Manager) on the Virtual Machine, The VM will be provided by the Customer on their ESXi license 5. Perform Standard & Advanced Configuration of Cisco UCM 6. Implementation of 30 qty. IP Phone 7. Rack Stack, & Power of New Voice Gateway Routers ISR4331-V/K9 (Qty. 1) 8. Initialize Voice Gateway Router • Perform staging & Software Upgrades • Configure OOB management 9. ...
WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk
Integrate Whatsapp API () on Vicidial - the freelancer will be responsible for sourcing the API. Please note we will only pay a one time procurement charge for the API and no recurring monthly charges API will be accepted. Please message for more details. Prefer a person with Indian tech connection.
We installed goautodial v4.0 from iso Kamailio running HTTPD OK SSL certificate OK RTPENGINE Ok Our main issue is the following: 1) Agent need to press (Login to dialer 2 times) 2) Can't register GoIP gateway (SIP Trunk) 3) Can't hear any voice. Only Goautodial V4.0 specialist is required...!!