Hey, guys, I have a simple project. I have a FreePBX server that I need to build an inbound Sip trunk to 2 separate carriers + build inbound routes for them. It should be a simple process. Please respond to the bid with "What up Dingo" at the beginning of your message so that i know you have read.
I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly
I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly
Need sip calculator script written in php
I need a new website. I need you to design and build my online store. Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, busines...Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, business loan, gold loan, car loan - investments, fixed deposits, insurance, mutual fund, SIP.
The business is a provider of cloud based communications which includes: Skype For Business Microsoft Teams Cloud Contact Centre Office 365 Sip Trunking Hosted PBX We require assistance in developing and revamping our website content, feature guides, product bundles, resources for customers (How2Guides/FAQ etc) some of which we have but need updating
I have all details .i need to configure sip and configure asterisk server . VIP experts
...security, quick SIP registration with a proxy, easy call routing management, multi-tenant PBX for each user sign up and billing system ready for online payment 3) Integration with CRM. Technology expertise 1) The engineer must possess good knowledge of Asterisk and Kamailio system 2) The engineer must have an in-depth knowledge of SIP layers 3) The
Hello i need to bulid sip filter to filter Retail Voip traffic
We are looking to develop a mobile app that will integrate specific API from our middle ware platform. Initially the app will only need to be for android devices. The mobile will need to integrate with a telephony platform for sip communication and a messenger platform. The majority of the mobile devices will not have cellular service and will be strictly
Looking for assistance in configuring Cisco call manager for sip trunking.
I need a very simple customisation of a Android based free / open source SIP client(sCIPsimple,Linphone or anything). I want the default values of my server to be populated in place of blanks in configuration forms. So, all you need to do is replace the values and compile and give me.
Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
...initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for the subaccount) 3. Historical reporting - whatever twilio provides data-wise on the SIP trunk side of the house (call date/time, ANI, duration
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any
Basically converting BICC to SIP and SIP to BICC using OpenSS7 or any other proved open source library. Performance and stability is an important factor
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards
I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step
I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on Android and iOS app. I would like that the cost of project be showed in separeted Android and iOS and hope that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features
I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows and Mac. And in the future on mobiles.
...experience in this field , this is simple problem. This is my result to run my code. ***********My code************ from pycall import CallFile, Call, Application call = Call('SIP/flowroute/18882223333') action = Application('Playback', 'hello-world') c = CallFile(call, action) [login to view URL]() *******************result******************** Traceback (mo...
hello, community I am looking for an engineer in networks for making a set up of an extension in a sip server remote.
dear all , i have pcap file which it capture from wire shark for my sip switch .. i want such script which will locate pcap file , and extract audio data from it .. it is RTP ... but i do not know the encode format (g711 or g722... ) the one who can have this project should first extract audio from sample i will send to him then we can start
I need build a sip app with vpn, sip and ssl, please contact for detail
...если все правильно и звонок пошел, то ему вылетит окошко, что вам звонят. Скрипт формирует USSD команду, исходя из введенных данных, форму которой я вам дам, логиниться на SIP сервер исспользуя Логин,Пароль и адрес сервера и отправляет комманду на сервер(делает вызов), отправляя эту команду происходит звонок с подменой номера, как только трубку поднимают
Company is looking vibrant developer who can build a SIP stack on a currently developing application. This facility will run on an existing app that is only available on Andriod. SIP stack must be designed to perform on single task. 1) Press a button to dial a fixed number on a SIP network.
...project named "VIBER WHATSAPP WECHAT FACETIME SIP GATEWAY", that you have finished it 90% can we talk about this and maybe you can develop this for me also? Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeCh...
Basic router configuration (IP Address, hostname, access credentials) SIP Trunk configuration from Cisco Voice Routers 1 and 2 to Genesys Virtual IP ISDN PRI ports configuration Dial Plan configuration Inbound and Outbound Calls Testing
Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat and Facetime to complete the call to the called party number. The development platform/operating system is not important. The project should be completed
...tarjeta SD - Diagrama de conexión entre la raspberri pi y el telefonillo (speaker, mic, bell, DoorOpen) - Indicaciones necesarias sobre cómo añadir nuestra propia configuración SIP a la Raspberri....
I need Sip Dialer With Include OpenVPN. Sip Dialer connect by OpenVPN. OpenVPN Connect only Sip dialer..... you need use PJSIP OpenSource SDK for SIP Dialer. I need Very Simple Apps... i need support g729 voice code. I can you OpenVPN server side. if you have good experience please contact me.
We have A2billing server and we are building an app for our resellers to topup customers through app. We need the working API to Create and Manage custo...are building an app for our resellers to topup customers through app. We need the working API to Create and Manage customers - mainly Create customer with Callerid (including sip account) and topup.
download elastix pbx and connect it with sip and Asternic Call Center Stats Lite
I need a website where I could sell mutual funds and insurances to my clients, and post regular information about the best funds and insurances. Also site must have SIP calculator and loan calculator. I need it preferably on wordpress.
Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP
I'm looking for programming help with [login to view URL] and Kamailio. Must have experience with SIP and preferably experience with [login to view URL] library. Will need 1 - 4 hours per week for 3 - 5 weeks.
We would like you debug and fix our sip trunking /Ext to Ext calling setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization, Elastix MT.3.0 has been installed already just need to fix some bugs. We are currently able to have our handsets receive registration, and our
My Name is Fayssal Daoud, my company TAMQEEN works as a business consultant with small and medium size enterprises automate their workflow to render them more efficient and profitable. I am currently working on an AGILE CRM project for an Auto Repair Workshop in the UAE. The required job is as follows; 1- Design the telephony integration optimum system 2- Generate a block diagram and BOQ of requir...
...(administrative network). There is a firewall that prevents connections from 4.0/24 to 5.0/24 We have VOIP SIP Server running at [login to view URL] Currently the server runs well but we will like to make possible users on 4.0/24 to access the VOIP SIP Server If you have long experience working with this type of routers and you know how to make the required
KAMAILIO SIP + MEDIA PROXY WITH SIP CAPTURE INTEGRATION INSTALL INSTRUCTIONS + CONFIG FILES + TECHNICAL SUPPORT We need a KAMAILIO CONFIGURATION accepting calls from one NIC with public IP address and redirect passing the call to 3 different SIP providers connect to 3 different NIC’s with “local” IP’s (load balance - round robin). Media must be redir...
...informações de cartão de crédito. Analisaremos como será feito durante o desenvolvimento; Opcional (projetado somente para atualizações futuras): Ligar para o profissional (Chamada SIP, somente na segunda versão) *Funcionalidades – Aplicativo Profissional* Login; Logout; Checkin; Checkout; Envio de fotos de atendimento com...
I am looking for a Asterisk developer to quickly update existing Asterisk code for dialer, inbound/ outbound/ press 1, and other advanced features. Need to configu...Polls AMD (Answering Machine Detection) DNC ( Do Not Call) Support Inbound Campaign Custom Caller ID Multiple Trunk Support Realtime Campaign Management Multiple Technology (SIP) RESTAPIs
I need a SIP cordova plugin to use with Ionic 3. Cordova plugin is already exist. [login to view URL] All you need is the create ionic plugin wrapper for cordova-plugin-sip.
...are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers, the upstream CDR reports are not suitable for processing the clients billing for
rebuild asterisk server using backup files sip and dialplan database AGI (python) restore all default functionalitys
...migrated to a Cisco 3945 router and it does not work fully. I have: Cisco 3945 with Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over
We have twilio and need someone to seup the Elastic SIP Trunking and make the phones dial out and ring... via Ciscso SPA540G and obihai
Hi guys, I have an android app which is already completed. It has few bugs on few phones. Can you resolve them? Are you aware of SIP protocol?
i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.
