I need someone who is an expert in terms of implementing VOIP services in websites. Its a small task with a strict deadline. Please bid if you seriously know how to implement VOIP. Deadline is 10 days max. All the backend has been implemented already. See you in chat.
...Telecom solution. So in asterisk that can do everything we want. For example, centos + asterisk + php + nginx + mysql + a2billing + freepbx multi-tenant and other service components to the server must be established. To run mysql on a separate server for the web site hosting services on the server will work on a backup basis. Asterisk on a separate server
I want to build a VoIP telecom company. We spoke yesterday. I think asterisk is the best for the job. all the properties of the Asterisk server to be managed via the panel I want. like for example the combination of a2billing and FreePBX. How do we work ? The long-term mean is constant if I want to work with a person who knows. Please tell me what you
The goal is to establish an asterisk server for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if
I need android application for door entry system. It should be built using SIP and VoIP. Design(include XML) was completed already. Don't bid without sip and voip knowledge, please. I will reject that man immediately. While chatting, let me provide design.
A STUN/TURN server has been tested to work on Android apps such as Zoiper and SessionTalk (using accounts from a specific SIP server). However, our app fails to use the STUN/TURN server correctly (with the same SIP server) and produces the following error in the STUN/TURN server logs: "error 437: Mismatched allocation: wrong transaction ID" The task is to resolve this error and ens...
I have freepbx on local machine connected to SIP at Twillio. Outgoing calls are working, Extensions with IP phone and soft phone work, but the incoming needs to be ...'INVITE' from (callid: firstname.lastname@example.org) - No matching endpoint found Also extensions are not showing connected or the IPs when showing peers in asterisk.
...#Skills&Qualifications: -Good knowledge on server administration on Linux (Redhat, Ubuntu, Centos) LAMP; -Understanding of VPN (OpenVPN,IPSEC); -Good knowledge of Nginx, Apache, MySQL,Asterisk; -Ability to write scripts(bash); -Understanding of Nagios, Zabbix and etc. #Must-have skills -Maximum attention to detail and analytical thinking -Orientation on research
Accounts on a SIP server have been tested to work with a STUN/TURN server on other Android apps such as Zoiper and SessionTalk. However, our app fails to use the STUN/TURN server correctly with the same SIP server and produces the following error in the STUN/TURN server logs: "error 437: Mismatched allocation: wrong transaction ID" The task is to resolve this error and make sure th...
Unifun is a provider of high-class IT products for Mobile Operators. Founded in 2005 in Moldova, today we have offices in 12 countries. Since 2012 we have been actively growing on the international market. During 5 years we began our collaboration with 95+ mobile operators. We have launched 200+ projects, with an average of 3-5 per month. Currently
I need a software developer to improve the an existing online portal ([url removed, login to view]) I am seeking someone creative who can find ways to make the website more efficient, attractive, and user friendly. You must be able independently Identify ways to improve the portal and to complete given tasks on a timely manner. You need to be able to maintain regular communication and keep me ...