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
This is a simple Webhook we need created so that when an SMS text is sent to a number within out Telnyx list of numbers we are able to see the contents of the text. We would prefer that the text is forwarded to one central number, but we just want to be able to see the contents of the text at the least. Please submit your offers and what your experience
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 : [kirjaudu nähdäksesi URL:n] regards
Needed a guy from Norway who has an access to mobile phones for 2-3 local mobile operators. We send the SMS message from our system and it is needed to check that the message has arrived at the mobile phone (screenshot it). No links, no need to open it - just to tell whether the message was actually delivered or not.
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) [kirjaudu nähdäksesi URL:n]() *******************result****************...
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
...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, WeChat
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....
...doesn't which has to be rectified. Also I have SMS OTP verification working fine earlier but from past few days, it is not working while I checked and the api from the sms provider is working but only it is not working from my website which also has to be rectified. I will provide the files in which the SMS api is added which has to be solved at urgent
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 [kirjaudu nähdäksesi URL:n] and Kamailio. Must have experience with SIP and preferably experience with [kirjaudu nähdäksesi URL:n] 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 [kirjaudu nähdäksesi URL:n] 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...
Plugin to send SMS campaigns within the Interspire Email Marketer: Create a SMS campaign Use custom fields Link size reduction SMS open tracking SMS campaign link tracking SMS bounce messages SMS reply messages Store replay messages Send SMS from contact list Use SMS on triggers/autoresponders
Hello, our website is [kirjaudu nähdäksesi URL:n] based on wordpress / woo-commerce. We need to integrate SMS API so that our customers can receive messages at various stages of orders. We already have sms gateway purchased which has given us their API to be linked. We Want you to link that with our wordpress site API Code: [kirjaudu nähdäksesi URL:n]
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
Hi, need SMS Application to send Bulk SMS (without any provider - free SMS) by Python to send Bulk SMS from the computer by uploading worldwide mobile numbers, with its names and sent to 5,000 mobiles daily for free SMS without fees and without providers. budget: 100 USD
I need a SIP cordova plugin to use with Ionic 3. Cordova plugin is already exist. [kirjaudu nähdäksesi URL:n] 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
...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
Hello, we are an SMS company and we want to know how we can make money over the sending and receiving of SMSs from our clients round the world, we want to make money from it and don't know how to go about it,i think I would meet someone here that would prepare a full doc on how money can be made, then once the doc is prepared, we confirm it and the
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.
I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".
...should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [kirjaudu nähdäksesi URL:n]
We are a Residential HVAC contractor that offers annual maintenance plans. We want to install the temperature and water sensors in the residential HVAC system that will trigger an alert/ Phish notification through an installed hub via text when to temperature split drops. This will make us very proactive in our services.