We need complete solution to run our gateway for voip termination. We have GOIP-32 port gateways and we need to run on puppy linux. So first we need to compile puppy linux which can be installed on PC and then all gateways will be connected to PC through switch or hub. web based panel we need for dial plan and routing. where we can see live calls, statistic, we can add client who is sending us traffic. We need also some advance options for anti blocking that can be discussed once we award this project. only serious people and relevant people discuss it. if you dont know and if its not relevant to your expertise then dont waste your and our time
We require Kamailio and Freeswitch integrator who can help setup a full unified communication for a startup. A training and handover documentation will be required at the end of the project. Someone who has done similar project will be highly preferred.
Hi I would like to configure Kamailio or OpenSips for load balancing of some freeswitch servers and I would like to use ASTPP as billing for that system. I would like to have about 1500cc Thank you
I am looking for someone to build out a Session Border Controller for my Hosted VoIP solution. I currently use Natpass, but am looking for a more open solution. You must have built out this configuration in the past. I do not want to pay you to learn. - SBC handles SIP Proxy - Registrations - Trunking - Security by locking down only domains we add I will need to see what experience you have before I hire you. I would like also to do an ongoing support with the person that builds out the system. 24/7 support, we can negotiate a monthly rate. Thanks!
Seeking to build, own, host, everything for a white labrl VOIP solution focused towards a specific industry along, used internally for an ISP and sell publically online through master agents. We need to hire multiple people for this platform who have expert knowledge of the various components. if you are one of them, please explain your areas of expertise in your bid and we will discuss from there. Task: - Installation of Kazoo - Scale Planning - Automation - Hardware - Configurations for Deployments - API Integrations with 3rd parties - Develop custom features - and more!
Hello, Comsys voicemail applicaton is widely considered to be one of the best voicemail applications/platforms. Please see Comsys website - voicemail description, here: [login to view URL] Unfortunately, their pricing prevents many start-up service providers from using it. Your job is to study Comsys features and use open source software to duplicate those features on a different software application. If you are knowledgeable about this subject, then you know you have 3 or 4 different options. If you have any questions please feel free to ask.
To build application fora fusionpbx project: Integration with crm (click to dial, pop up, instant messaging) We are starting with a vloud pbx project based on Fusionpbx. We wwill sell tenant to our customers. We need to improve the features concerning CTI,API,mobile integration, conferencing etc. Integration with Office 365- Skype for Business (calls in/out via fusion pbx trunks) CTI with instant messaging for freeswitch-Fusionpbx with extensions/status, LDAP, instant messaging, call logs. The cti need to be improved with SIP videocall features and conferencing up 3 parties. Videoconferencing application to manage videoconference rooms up to 6 parties nwith moderator, invitation etc...
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 … - [login to view URL] - [login to view URL] Where each subdomain has different ip whitelists and also help us setup graceful deploys where we can take Freeswitches out of circulation to push changes without dropping calls by draining live connections and swapping in standbys its a pretty big project
I need help to create a trunk between Freepbx and my other SIP server. There are E1 cards connected to FreePBX and we assign DIDs from Freepbx to end users. They register by ID and Password to receive incoming calls. We don't have any issue there. But now we have another request to forward bulk DIDs to another SIP server and the authentication will be only IP without any user or password. It is simple project which should not take more than 1 hour to complete for anyone who knows how to work in FreePBX.
- Mensagem de Bem-Vindo - Mensagem da Administração - Salas Gerais - Salas Privadas - Salas Numeradas - Sala da Monitoria - Pin - Torpedo - Lista de Participantes On-Line - Murais de Mensagens - Caixa Postal - Menu de Opções Básicas - Menu de Torpedos - Menu de Opções Avançadas - Menu Opções de Pin e Caixa Postal - Menu Trocar Senhas ou Gravação do Nickname do Pin - Menu de Caixa Postal - Menu Lista de Participantes On-Line - Recurso Salas Numeradas - Recurso Sala da Monitoria - Menu de Murais - Menu de Opções do Mural - Recurso Regras do Serviço - Enviar Uma Mensagem para a Administração - Cadastramento de monitores Obrigado!
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
I have a fresh installation of ASTPP , i need the following configured: 1. create origination carrier ( customer) 2. create termination provider [login to view URL] 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 customer and route his calls to trunk of termination carrier. test and make sure all calls connect properly. give me a walk thru of the steps taken to do each ,.
Dear, All Viewers we are looking for someone who can help us to create Asterisk Native Dialplan Using ODBC to control callflow according to our logic without using any AGI, just using asterisk pure dialplan. Our goal to make lightweight fast execute & 0 resource using, please only experienced people bid, waiting for your bids, Thanks for your attention.
We need Push Notification feature via Event Based Routing in OPENSIP. Please BID if you have knowledge of OPENSIP.
viciial server setup and installation on centos with configuration for calling . need to start today and assist in installation and configuration. Thanks
We have a cloud based phone system running FusionPBX. We are starting to have a large number of phone / system orders, and need to setup provisioning correctly - so we can drop ship handsets to a customer site, and then the correct details get provisioned to the phone. We need to build provisioning templates for our standard phones (Gigaset N300 / N500) and a variety of handsets (S650H, CX430H etc.) We also need SNOM, and LG / Nortel IP8840 as well. We would also like this functionality for our broadband routers as well.
We need create gate call forwarding from SIP to some messengers Viber, Skype, Whatsapp, Telegram if it possible. And send messages to these messengers. ------ Необходимо создать гейт SIP в мессенджеры Viber, Skype, Whatsapp, Telegram (в какие возможно). А так же отправлять в эти мессенджеры (плюс facebook messenger) сообщения.
We need a specialized consultant to install a FreeSwitch server in the Microsoft Azure cloud with the following capabilities: 1. Support for video conferencing with Vertor Communicator; 2. Support for secure communication with TLS and RTPS; 3. Communication to SIP Trunk using [login to view URL] as provider; 4. Graphical interface for creating extensions; 5. Possibility of using database for creation of extensions; 6. Installation of Portuguese language for all functionalities. The project should be built together with our technicians to be trained during the implementation of the environment.
there are a few different locations using LAN, some of them using VPN to connecting, need someone who could do the LAN, VPN maintenance, we also use quickbooks. Need someone who is also very familiar with quickbooks. Chinese speaking would be plus
I'm integrating a2billing to my asterisk platform, this system will require me to bill the recipient or the callee for receiving calls. Below is the detail scenario of my application: 1. UserA and UserB both register and connect to my asterisk platform with their extension numbers respectively. 2. Both of them also have profile on the a2billing platform. 3. UserB setup a DID on the a2billing platform with his GSM mobile number as the destination for the DID. 4. UserA calls the userB DID and the call terminates on UserB GSM mobile line via a SIP trunk to PSTN. 5. UserB is billed for the DID to PSTN call. 6. UserA is not billed for anything. The above is what I want to achieve. I've been able to do the setup, but right now, only UserA is being billed for both A-Leg and B-Leg of the calls. But what I want is for only UserB, the owner of the DID to be billed for any calls to his DID that terminates on the PSTN destinated GSM number.
We need a skilled sales representative who will support our company in selling and buying VoIP routes. Voip experience highly preferred. Perfect written English, Above average spoken English needed. Experience with CRMs.
Hello! i have a customer who recently got hacked his Elastix PBX so he need us to develop the task to make it to run again, then while i will be checking the elastix contexts, routes and the core PBX to check if everything it is ok the customer wants also other tasks to be performed like: 1.- Make the Vtiger CRM to run again (apparently it is offline) then needed to be running and up (consider maybe the customer can ask extra things of the CRM like how to use it or things like that) 2.- Make the Owncloud to run again, apparently it is offline and they want it to run (i dont know for what they are using it) 3.- Help the Elastix to run again the callcenter campaigns (maybe we need to make one for testing purposes or something then consider it also)
I need someone to configure-test the first trunk in a2billing and fix access to the asterisk-gui. I have installed the asterisk-gui but can't access it over http for some reason. I need someone who can get this fixed super fast. I have ran the command "make checkconfig" and see that all is good. I have even set the bind ip to mine but still can't access the gui. Maybe Marian db issue or wrong port setting in [login to view URL] I have tried a few things. It says everything is ok so it should be working. If you know what your doing in Asterisk and a2billing this should literally take you no more than a few minutes. Please only reply if your fast and efficient. We are also looking for someone for steady work if you prove to be fast, reliable and honest. I am a server admin (just new to asterisk) I have installed this myself so i kind of know what is going just not how to fix this issue.
Total budget $5, Installating OpenVPN on VPS and give me guides. Net pp2p or ltp protocol for configure on tplink router.
We are trying to get customization to PJSIP source code to be able to do transferring using PJSIP within Asterisk. I have a requirements doc but need to interview anyone interested in the project before I send it to them.
We are developing a unified communications platform, which will operate from a number of sites globally. We require a resource to develop the VoIP Services platform (VoIP back-end) that will provide all call handling functionality for the platform.