Need someone with openSIPS/freeswitch experience the initial task would be to help us setup and configure a high availability deploy of OpenSIPS ->[ Freeswitch1, Freeswitch2, Freeswitch3, … ] the other aspect is I want to setup OpenSIPS to handle … - [url removed, login to view] - [url removed, login to view] Where each subdomain has different ip whitelists
We are looking for a Freelancer with good experience in OpenSIP configuration with Freeswitch in a fail-over/ Load balancing environment with session recovery. The Freelancer need to work through Team Viewer and need to provide documentation on Environment/Prerequisites and configuration performed for a successful delivery of project.
Hi Guys, I need a an opensource sip platform which can enable video calling and audio call conference with Mizutech webphone. Webphone works with webrtc so the sip platform must support webrtc. Payment will be released only after successful testing.
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.
We are looking for someone with a good understanding of kamailio and freeswitch. We are currently building a strong reliable VoIP cluster. The next step of our project is to try and fix nat issues, and to protect and load balance our media servers. Someone with strong knowledge of kamailio, and can coach us so we are using kamailio the best we can
Hi, Looking for some one who can teach installation and customization of Asterisk freeswitch a2billing kamailio all in standalone server and connecting of writing application
Hi rawat270, I noticed your profile and would like to offer you my project. We can discuss any details over chat. I have ASTPP. I need to modify the callerid that the customers send and set a new CLI to all calls from them. Are you able to do something like this.
...web based gui, web based gui maker, google scraper web sites, web based gui asterisk dialplan, net scraper web page, designing web based gui, screen scraper web crawler, freeswitch web interface gui, export parts site scraper web, screen scraper web page excel, scraper web access, web application gui, web ajax gui, asterisk multi tenant web based gui
...of day, inbound phone number and user input. OUR COMPANY Philosophy At RingRoost we strive to take the complexity out of VoIP technologies. Software like Asterisk, FreeSwitch and FreePBX are great tools for companies running on VoIP, but are still only a small part of the toolkit needed to properly service businesses and VoIP users. After years
...publicate a API for my platforms in Asterisk or FreeSwitch (I have both platforms) I think use a open or free api platform, to simplify the developers. Examples: [url removed, login to view] [url removed, login to view] [url removed, login to view] This API will be use to make calls through Asterisk and FreeSwitch servers. The API will use to send calls
Hi there. Looking for a high-skilled professional in Python/Django/MongoDB to combine all the things together & create work...interface post write text file, write web based interface administration , python web interface, web interface python django, write app interface with third party web portals, freeswitch web interface, web page design work
Hi. I am new on ASTPP. I use the link([url removed, login to view]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE:
I have a fresh installation of ASTPP , i need the following configured: 1. create origination carrier ( customer) 2. create termination provider [url removed, login to view] origination rate table and termination rate table 4. create trunk and set up routing so that origination carrier calls cam be terminated to termination end. 5. create a sip user account on new
I am planning to set up a company for Hosted PBX and Call Center. Based on Asterisk/Freepbx /Freeswitch etc. I would like to set it up in Amazon AWS so that we do not have to worry about the servers. Will have SIP trunking for the incoming and outgoing calls. Each customer will have their own portal to manage their users, extensions etc which
Hi Aqs Y., I noticed you have a fully developed ringless voicemail platform. Can you send me screenshots and test /demo access? Would like to know price and...you have a fully developed ringless voicemail platform. Can you send me screenshots and test /demo access? Would like to know price and if this works on which voip service freeswitch, twilio etc