VOIP SWITCH CONFIGURATION AND INTEGRATION TO WEBSITE We are a Voip Telecommunication Company, offering mainly international calling solutions to our customers. Our main services include, Pinless Calling Cards, Wholesale carrier services, International Mobile Top-Up, Voip Residential and PBX solution. We have just migrated our customers to the voipswitch
Hello, Adapt a Twilio Application for Voip Innovations, a competing provider. [url removed, login to view] You can use the following Voip Innovation's documents: [url removed, login to view] [url removed, login to view] Please contact me with any questions. Thank you.
i have local server with pri card, i have DID configured through the PRI card , now i need to move the local DIDs to cloud asterisk server and configure there sip accounts to receive the calls and setup ivr there.I am willing to pay 50$ for this project ,please do not bid if you think this amount is less.
OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO
Asterisk - CTI (Click To Call) API This will work on a Asterisk 13, FreePBX 13, and Asternic Pro 2.1.x platform. We need to provide a REST API for several actions e.g. a) Click To Call - e.g. /[url removed, login to view] authid = preset hash value to authenticate the user against. Return in JSON - Asterisk
...syncing for both server A and B. If server A goes down server B must take care of all the things. Like MX records DNS entry. 3. We have a newly installed Asterisk server. Need to configure the Asterisk server. Here are the following things you need to do. i). Configure all extensions. There are total 7 extensions need to create. Those are given below
We are looking for VoIP expert to join our noc team for monitoring and analysis of our current routes, you must be have previous experience. You will be dealing with VoIP routes, identifiying bad numbers on the routes using the CDR and Human behaviour data and call filter, adjusting the dynamic call filter and Human behaviour server.
We have all ready own [url removed, login to view] first all comments are will be clea...mobile telephone number is all ready in database say sorry and if email is all ready in database say sorry 7- mysql database name wsip 8- to create sip account will use asterisk database 9- delete account user we must delete from [url removed, login to view] if success than we delete useraccount
I need an experience free lancer that can fix connectivity issues and Vicidial asterisk based system. If you don;t know Vicidial please don't even send your proposal. Long term project with some support needs on demand.
I have a VoIP and web design project that requires the worker I hire to be skilled in: VoIP – This includes experience with VoIP phone system features and functionality, VoIP phone service provisioning, IP phones, VoIP Billing systems and all VoIP terminology. Web Design – Skilled in web design such as Dreamweaver. Knows PHP and ecommerc...
I require to develop VOIP (voice over internet application) on preferably android platform, but windows OS is acceptable. Project is not meant to be done from scratch and I expect freelancer to be adept at utilizing already available open source projects to drastically cut development costs and time only modifying already existing solutions to my needs