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.
...on two different servers and the same information is supposed to be there. So made redundant. On the asterisk side in the same way, we want a redundant structure. In terms of VoIP services, there should not be an interruption in service. We work with a global Tier 4 certified data centers. So continuity is very important to us. The following is my personal
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...
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...
OTT MOBILE Dialer is customizable mobile OTT solution that allows Communication Service Providers to build their own branded VoIP based OTT app and launch service on the existing network. Users registered under Service Providers' brand, Service Providers can fight the rising competition from top OTT players like Skype, WhatsApp, Viber and others
...both identify and fix the issue. There are a couple of other configuration issues that I need fixing as well and I'll provide a list. The issue are such as setting up the PBX to forward calls to either voicemail or a mobile phone depending on the state of service (e.g. night service etc). Nothing tricky, just a very basic set up for a 5 extension
...the open source Linphone Flexisip to interact with our Linphone IOS mobile app so voip push notifaiction can wake up app, to receive call . Accounts handled by Portaone Voip Switch . Flexisip config done, voip push certificate in place, still some issues not allowing voip push. Trying with another programmer as well, but still we are not quite there
I have just installed a Kazoo VOIP server and I am having difficulty uploading the rates through a csv file. I need someone with expert knowledge is Kazoo to help me upload the rates and show me the procedure for uploading same. Also I would like you to put me through the process of creating the service plan. Please don't bid if you are not an expert
...please suggest other means. We need to keep our costs down in what we are charged to send through which ever provider is used here. We are willing to setup a server (if Open PBX is a possible part of the solution) or whatever. The most important part here is the MP3 must go directly to voicemail. Prefer to have code in .NET, but if you can get it
We need VoIP Developer to implement cross platform audio calling facility into our mobile applications and Web browser.
Fix the following error seen in two different TURN servers 'Coturn' and reTurnServer causing failed SIP call: Coturn log: session 001000000000000001: realm <> user <>: incoming packet message processed, error 437: Mismatched allocation: wrong transaction reTurnServer log: DEBUG | [url removed, login to view] | reTurnServer | RETURN | 139914911389440 | [url remove...
We have .NET Restaurant Management System. this system has a ...have .NET Restaurant Management System. this system has a module which handles clients' orders through the phone. we Have Yeastar PBX (different models) which is Asresik platform. we need to connect our system with PBX. So once incoming call happens a pop-up screen appears with caller-ID