I have a small office with multiple Yealink T-46 VOIP phones. I'd like to setup a provisioning server per Yealink details so these phones can automatically download their configuration when I want to make changes rather than having to change each phone individually. I can host the server, just need someone to figure out the setup.
Problem is that the dependencies don't install using sudo apt-get in the live cd version of ubuntu 17.04 Actually terminal displays package not found for sip and for pyqt-4. I need you to write in your bid how to achieve before 24 hours.
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
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 tweaked. I get error: NOTICE: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from (callid: firstname.lastname@example.org) - No matching endpoint
Sonstiges oder nicht sicher Sonstiges oder nicht sicher Quiero hacer un proyecto de un videoportero IP En Raspberry y la llamada debe ser a un android o iPhone vía SIP video allí. Lo quiero construir para fraccionamiento. En un Touchscreen seleccionar el departamento/casa/persona y Raspberry lo hace el vídeo allí al celular/ tablet.
Need a designer to create branding and logo for a new coff...for a new coffee shop as well as packaging and tshirt design for employees. Looking for a very clean looking Logo using grey, white or black colors only. The Cafe's name is Sip/Bite. Attached is the initial logo rendering, but we are looking for new ideas and designs (very modern and clean).
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
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
I have a panasonic NS700 which will accept incoming calls but I can't figure out how to set up outbound calls. I have 2 trial trunks, one with Sipgate the other with Gamma. I need tomeone that is very familiar with the Panasonic NS500, NS700 or NS1000 to look through the settings and both identify and fix the issue. There are a couple of other configuration issues that I need fixing ...