I am looking for a freelancer to help me with my project. The skills required are Android, Cisco, FreeSwitch, VoIP and Windows Phone. I am happy to pay a fixed priced and my budget is Rp2500000 - Rp7500000 IDR. I have not provided a detailed description and have not uploaded any files.
HI, I have a voip switch running on windows. I am open to change the platform to windows if required. I am looking to use a mobile dialer as client with encryption (zrtp or srtp / TLS). I have found few dialer on playstore that have the option of this encryption, however I need assistance in setting this on the server side. Only freelancers having
I will use a model and go through the research. Make sure that I will do the model and finish the expert and get the result so you do the analysis. We will work together. File will uplode after I discuss with you.
I have one static IP address. I need someone to set up my mikrotik routerboard so that the ports are forwarded from the default VoIP ports and set this router up so that I can access my sip server from remote extensions. After this is configured I need the router configurations to be provided to me via zip file.
Necessitamos integrar CRM versão 7 ou 6,5 com provedor pbx asterisk voip de modo que seja configurado para abrir tickets automaticos, leads automáticos de acordo com regras do provedor de pbx. O provedor de pbx possui regras de inserção de cpf ou cpnj para novos clientes e opções para chamar ramais das áreas comerciais, financeira, suporte, ...
I need you to develop some software for me. I would like this software to be developed for Linux using PHP. We are looking for FOIP/VOIP development which have high channel capacity and server redundancy. We would like to check demo of the solution if already built.
We're building a Cordova application, and we're integrating with Sinch VoIP APIs ([url removed, login to view]) They do provide a web sdk but we couldn't manage to use it for receiving a call when the app is closed. So we need a working proof on concept for receiving a sinch call while the application is closed. It is up to you to build a solution that utlize the
...dials the number to the Customer (+/- callback) The program - must select two numbers for the call, it is important that we can use our own SIP account / SIP trunk We have a server VoIP from [url removed, login to view], the platform has almost all the functionalities that interest our clients, in addition to the one mentioned above, that's why our inquiry is for you
IT Support – Technical Engineer – Windows – Microsoft – VOIP – Network – LAN WAN – Active Directory – Technical Consultant – Help Desk – Service Delivery - Remote Support Field Engineer Candidate is required for remote support via remote desktop to resolve issues with business clients. We Cover approximately 20 business clients ...
...project production. At minimum an Initial phone conversation before start and phone/voip conversation is required to start milestones and payment. Due to the nature of this project, all features must be tested and working on my end when I install the code on my live server/device before final payment is sent. Begin all quotes that you prepare replied to
...companies looking for a new telephone system. We offer a competitively priced Hosted VoIP system. Money held in Milestone until completion. The project must yield 100 qualified leads to be deemed completed. Criteria: Willing to move to a cloud based VoIP phone system Minimum spend of £200 a month on current call costs or projected Must
We're building a Cordova application, and we're integrating with Sinch VoIP APIs ([url removed, login to view]) They do provide a web sdk but we couldn't manage to use it for receiving a call when the app is closed. It is up to you to build a solution that utlize the web-sdk or their native sdk or any other hack as long as our test case passes (the 3 test cases
Hi, I need VoIP Business training - What i basically need to learn is Centos and Asterisk and how it works I also require trainer to teach me mysql and How to create Database on DBFORGE to access server and makes entries. The trainer must be in uk Thanks
We have several SNOM M25 and M65 Phones connected to a SNOM M700 Base. The Snom uses FreePBX Asterix 14.0.x (latest) as SIP-Gateway via PJSIP connected to SIP Gateway of German Telekom. All the Phones are properly connected and the Peers registered. Somehow, we can't manage to dial out or to be dialed in and we don't know why. Need an SIP Expert to deal with this and configure the system...