...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
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: email@example.com) - No matching endpoint
...#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
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
...Skills&Qualifications: Good knowledge on server administration on Linux (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
...below. in app purchase. ● Web Dashboard ● Users management. Search, Create, Edit, Remove, Charge balance, Groups ● Gateways management. Set termination paths, SIP trunking, Set Prices, Set countries ● Billing. Multi currency support, Minutes based charging, Promotions support. ● Built in XMPP server, Instant messaging, Files transferring
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 ...
When calls come in, we want the ivr to say "The number you have dialed #### is on the line, please hold while we try to connect you.". The #### is the number dialed and we need Freepbx to dynamically convert text to speech for the user to hear. You need to configure as well as teach us how to configure.
We are looking for long-term asterisk experts on hourly basis. We need very strong developers with excellent hands-on experience and debugging abilities. A senior developer with design skills and aspiring to be an architect. You will have to sign an NDA.
To configure 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. Do necessary edits on exi...