Softphones srtptrabajos

Se necesita tecnico experto para montar remotamente una central asterisk, que pueda hacer llamadas locales usando un modem 3g con chip celular. Configurar softphones para personal.

Necesito crear una VPN segura en USG UNIFI para acceder remotamente desde clientes windows a carpetas en NAS Synology y también poder utilizar softphones sip desde PBX Grandstream UCM61xx. Gracias.

Hola! tenemos una PBX Grandstream ucm61xx y el hard de red es ubiquiti. Quisiera implementar softphones con iax2 (no sip) para poder tener internos remotos sin comprometer la seguridad de la red.

I need a SIP Client programmed in python. This software should be/have: -Both, the SRTP protocol and the ZRTP protocols are needed. -It should have video calling and graphical interface (GTK ...). -I also need to have the ability to configure 2 extra buttons (relays) for different purposes. -I also want to be able to add a logo inside the graphic interface. -And lastly I need to own copyrights of this program.

Estamos buscando un profesional que configure una central telefonica, la cual debe correr desde una maquina virtual e...profesional que configure una central telefonica, la cual debe correr desde una maquina virtual en un iMac con Yosemite. La configuración de la central se tiene que hacer remotamente sobre la maquina virtual. El profesional debe poder brindar soporte del proyecto en un futuro. El hardware con el que se cuenta es: GXP1610 Small-Medium Business IP Phone (4) GXW400x Series IP Analog Gateway (1) Softphones en iOS y android. En la actualidad contamos con una solución que instalamos nosotros mismos con Asterisk to go, lo cual es muy sencillo y solo viendo los tutoriales se logra configurar, estamos buscando son funciones especificas relacionadas...

Hola necesito programador con experiencia en Protocolo SIP, el cual este familiarizado con Linphone SDK, PjSip o otras SDK el cual permitan el desarrollo de Dialer, o softphones para Android y Iphone.

Im recently starting my first Asterisk project. After install it, everything is already configured and working properly thru softphones (X-Lite), but Im completely stuck how to configure SIP phones as Polycom IP650 with my Asterisk. Looking for somebody to help me configure and learn how to configure those phones now on. I have a small IT company in Miami, starting installing Asterisk servers to some clients. Recien estoy empezando a instalar mi primer servidor Asterisk. Todo esta funcionando bien con los telefonos X-Lite, pero no conozco nada de LINUX para poder configurar los archivos, el TFTP Server y demas cosas que segun veo se necesitan para conectar telefonos duros SIP. Quisiera ayuda para configurar un ejemplo y aprender a como configurarlos en lo adelante. Tengo una pequ...

...veces. En otros momentos la llamada se escucha perfectamente. La troncal esta configurada, las llamadas salen pero , A VECES, no se escuchan bien. Hay que colgar y volver a llamar y al hacerlo , se escucha bien y a veces no. Es decir, el problema es que se establecen las llamadas pero , a veces, no se escucha bien. Las llamadas se realizan desde republica dominicana conectandonos a traves de softphones zoiper a la centralita Voip Elastix en Barcelona (España). El problema tendria que estar en la centralita en España. Llamamos sin problema por voip desde la oficina en republica dominican (via skype o conectandonos a una empresa de telefonia ip directamente). Por tanto creo que esta relacionado con la configuracion del portech o algun problema con este equipò ...

Hola, somos una empresa pequeña de turismo receptivo en Bogotá y queremos implementar un sistema de PBX virtual que nos ayude con la recepción de llamadas y el manejo de estas. Los requerimientos son los siguientes. Requerimientos básicos 1....El usuario debe escuchar una grabación hasta que le atiendan la llamada. 7. En las opciones 1,2 y 4, en caso de no obtener rspuesta después de X segundos de espera, se desvía a un buzón de voz con notificación a un correo. -------------- Requerimientos Adicionales 1. Sería ideal que al contestar una llamada, nuestros agentes sepan las opciones que ha elegido el cliente. 2. Si los softphones instalados en smartphones de las opciones 1,2 y 4 no están ...

Im recently starting my first Asterisk project. After install it, everything is already configured and working properly thru softphones (X-Lite), but Im completely stuck how to configure SIP phones as Polycom IP650 with my Asterisk. Looking for somebody to help me configure and learn how to configure those phones now on. I have a small IT company in Miami, starting installing Asterisk servers to some clients. Recien estoy empezando a instalar mi primer servidor Asterisk. Todo esta funcionando bien con los telefonos X-Lite, pero no conozco nada de LINUX para poder configurar los archivos, el TFTP Server y demas cosas que segun veo se necesitan para conectar telefonos duros SIP. Quisiera ayuda para configurar un ejemplo y aprender a como configurarlos en lo adelante. Tengo una peque&...

Quiero implementar un PBX en la Nube podria ser en AWS de amazon con lineas VOIP y extensiones TLS/SRTP podria decirme si lo vez viable ??

...C#, or any other compatible with the Milestone SDK. Tools: Milestone SDK, SIP libraries such as PJSIP or Linphone. Compatibility: The SIP client must be compatible with current Milestone versions. Support for standard SIP protocols and common audio codecs such as G.711, G.729, etc. Security: Implement security measures to protect the SIP client configuration and data transmission. Support for SRTP (Secure Real-time Transport Protocol) encryption and TLS (Transport Layer Security) for SIP connections. Work Plan Design Phase: Initial meeting with the development team to define detailed requirements. Design the architecture of the SIP client and its integration as a plugin in Milestone. Development Phase: Develop the SIP client and configuration interface as a plugin. Implement ...

Project Description: FreePBX and GOIP Configuration Looking for the initial setup of a FreePBX systems with GOIP GSM gateway for inbound trunks. Softphones will be used by internal clients to receive calls. Skills and Experience Required: - Experience with GOIP GSM gateways. - Proficiency in FreePBX configuration. - Softphone configuration. - Understanding of custom configuration setups, if necessary If you have the expertise in these areas, please bid on this project. Thank you.

I am looking for a freelancer who can help me set up and configure my Twilio account. Here are the details of the project: Setup twilio account for using soft phones Simple IVR menue Sip trunk Preferred Communication Channel: - Both SMS and Voice Call Workflow: - I have a detailed plan that I want...to implement with Twilio Technical Level: - I have an intermediate level of understanding with Twilio Skills and Experience Required: - Experience in setting up and configuring Twilio accounts - Knowledge of Twilio's SMS and Voice Call features - Ability to understand and implement a detailed plan - Strong communication skills to collaborate and finalize the workflow - experience setting up softphones/ sip routing If you have the necessary skills and experience, please apply fo...

...Server Skills and experience required: - Strong knowledge and experience in hosting VoIP softphones on Windows servers - Proficiency in configuring and managing VoIP protocols, particularly SIP - Ability to handle concurrent calls ranging from 1 to 10 Project details: We are looking for a skilled freelancer who can help us host a VoIP softphone on a Windows server. The server will be running on the Windows operating system. Key requirements: 1. Operating System: Windows 2. VOIP Protocol: We are open to suggestions as the client did not specify a preference. 3. Concurrent Calls: The softphone should be able to handle 1 to 10 concurrent calls. If you have a strong background in hosting VoIP softphones on Windows servers, and are proficient in configuring and managing VoI...

...functionality i need for a virtual receptionist which vonage business has (Ring to user before going to Virtual Receptionist & Call Tagging). Vonage does not have the outbound capabilities I need (Automatic Local Presence for Calls & SMS & a Power Dialer). So I'm trying to figure out a way to use both inside Salesforce, but am not sure what is possible. 1st idea is creating 2 separate open cti softphones within salesforce. One for all outbound calls (click to dial, dial pad, and power dialer) as well as all SMS features (360 CTI/SMS will have this part covered). The 2nd softphone (Vonage Business Center or possibly Vonage Contact Center) would handle all inbound calls so that I could have a ring first to & call tagging feature for the inbound call while ...

...Messaging with end-to-end encryption for both one-to-one and group chat Capability to share pictures and files Address Book integration Call History display Presentation of advanced call statistics Echo cancellation feature Quality of Service optimization Ability to send and receive SMS text messages through the ASTPP PBX server Implementation of secure communications through zRTP, TLS, and SRTP Support for Bluetooth headset devices Language options: English, Chinese Integration with ASTPP PBX server to check balance Integration with ASTPP user accounts Message Archive Management (MAM) functionality Implementation of PUSH Notification from the ASTPP PBX server (FreeSwitch) Customization of the app with our logo and company information Provision of source codes. K...

VoIP Recording - SIPREC Re-Invite and Audio...VoIP system and I need assistance with SIPREC Re-Invite and audio streaming. - I require live streaming of the audio for real-time monitoring and analysis purposes. - There is no need for transcription of the VoIP audio stream. Skills and Experience Required: - Experience with SIP VoIP systems and SIPREC Re-Invite is essential. - Proficiency in audio streaming technologies and protocols. RTP and SRTP - Knowledge of real-time recording and live streaming techniques. - Familiarity with VoIP monitoring and analysis tools. - Attention to detail to ensure accurate audio streaming without transcription errors. - The code implemented could be in .Net Core, Python or Java. - Knowledge of Acme or Sonus SBCs, and Avaya, Cisco and Genesys is a di...

I need a webrtc to sip gateway to be implemented so I can connected some webrtc softphones (asterisk webrtc softphones on our Odoo CRM) to twilio sip domain

- Install fusionpbx - Setup Inbound/outbound SIP trunks - Setup 3 extns - Test softphones - Delivery

I need an Android app similar to Zoiper, for business use. The app should have basic call functionality, including making and receiving voice and video calls. Additionally, the app should have a modern design that is easy to use. The ideal freelancer should have experience in developing VOIP SIP softphones for Android.

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome to join too after going through a interview ca...

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

We are looking for an experienced VoIP developer who can design Windows and MAC desktop VoIP applications using our Hosted PBX API. The application will have to be tightly integrated with our asterisk-based PBX and our custom API. Supported functionality will include: Voice calling via SRTP Searchable Call history with access to call recordings and call notes SMS and MMS messaging Read-only access to favorites and BLF keys Read/Write access to personal contacts Visual Voicemail Do not Disturb Call Forwarding We prefer a web application running installable with an Electron wrapper on the client's workstations but are willing to entertain other options.

...fully working sip account after award) The caller-URL contains some query parameters, like the callers number. The app can run only one instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL. use jitsi desktop event handling The environment: - your SIP jitsi app - one or more SIP desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone The listener shall be configurable, means - what URL to call - which phone numbers to ignore (regex) - which phone numbers...

...caller-URL contains some query parameters, like the callers number. The app can run only one instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL. use or similar java library/framework The environment: - your SIP application to implement - one or more SIP desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone We have some exiting code of a proof of concept, which does not work anymore. So you do not start from scratch. But it is mandatory that

...SIP provider only. The caller-URL contains some query parameters, like the callers number. As a UI-less (cmd line only) app on the desktop it will be so a simply app which does basically a call listening and triggering a browser based URL to show. The environment: - one SIP account which has a real phone number - your SIP application to implement - one or more SIP desk phones - one ore more softphones (laptop, smartphone, ...) 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone We have some exiting code of a proof of concept, which does not work anymore. So you do not start from scratch. But it is mandatory that you are a master...

Asterisk full setup in our enviroment ( 10 user , 6 cisco phones). Full setup as a virtual pc on Synology NAS. Setup for softphones on mobile devicese and connect existing cisco hw spa504g phones + call recording etc ...

we are looking for a very experienced ...are a bad fit, if you have never done tasks of integration SIP into Java A simple test task for seniors in SIP+Java: - listen to incoming calls (SIP provider can be provided to you) - trace incoming calls to sysout with the phone number Milestones: MS1: very basic implementation of SIP MS2: the rest of the above named requirements MS3: with and without TLS / SRTP support TLS / SRTP, shall be mandatory for later usage of this app budget? will not be disclosed, place your best bid timeframe? starting in the next 2 weeks time commitment? about 1-4h a week for the initial consultancy. If you want to get incorporated into development, you will be awarded tasks, which you estimate, and we award you afterwards the implementation

...JUnit (no UI) What is NOT needed: - a UI (not required, implement a JUnit test to call your functions) - a service architecture (like spring or JEE) - any persistence - a voice/audio implementation for SIP (only incoming ringing required) Milestones: MS1: very basic implementation of SIP MS2: the rest of the above named requirements MS3: with and without TLS / SRTP support TLS / SRTP, shall be mandatory for later usage of this app What are our requirements? - your code passes checkstyle, pmd and spotbugs (we will share you a git repo with eclipse settings) - JDK17 - maven - 24/8 formula - create a model class representing the input of your function - create a service class implementing the logic - create a unit test, which tests the service class - we do NOT need

I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.

I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.

Dear Amrit, we changed the login configuration of the mobile application and this project is catered to support the work of this configuration. With the configuration, we will look at the app enabling SIP TLS and SRTP. We also added the domain setting so that user can change that domain setting. Thank you. John

i need help to setup my gsm gateway into one login id and password for softphone so that we can receive calls on the move

...including GUI. PBX shall be deployed on premise on server/hardware. It will be limited to number of users/phones and license to be uploaded/entered for upgradation of users/phones. GUI will have company branding. PBX software shall be inclusive of below modules but not limited to these only. - Detailed CDR - Call Recording - IVRS and log - Phone Logo change - Live Monitoring and real time status - Softphones integration - Video Conferencing - FXO/FXS/PRI/Trunk Gateway Integration & support - License based features to limit number of users/phones - Free Modules of freepbx/Pbxact - Built-in Redundancy Developer need to provide 6 Months support in this project from developer for testing/bugs/reports/problem of any kind. Future upgradation shall be possible on the software and s...

Configuration de 4 pfsense dont 2 en cluster et Configuration d'un Freepbx pour une utilisation avec softphone ( hors éventuel prix des softphones si non libre)

...aggressive goals and high-quality standards. Creative approach to problem-solving. Ability to develop long-range project plans and schedules. Experience in embedded software and firmware design and development. Bonus Points: Comfort with tools such as debuggers, logic analyzers, and oscilloscopes. experience in Embedded C/C++. Experience with network programming and protocols (TCP, UDP, HTTP, SRTP, etc.). Experience in working up and working with hardware-focused communication interfaces (MIPI, SERDES, I2C, RS232, USB, Ethernet) Experience in working with HAL Firmware development experience on MO, M3, or M4 embedded cortex Development in embedded C/C++ in IAR, Mbed, or Keil Development Environments Thorough knowledge of Embedded System Architecture with RTOS. Prior experience ...

We need a Technical person who can install a functional VICI Dialer on a server of his choice and knows how to configure softphones like Zoipher, eyebeam to the Dialer for US Calling. He will be paid for Installation as well as maintenance of the dialer very month. He must know about AC-CID and the Outgoing call must be that of the Customer. Must have expert knowledge of Installing VICI Dialer and Configaration

Install TLS 'let's encrypt' certificate, setup PBX to use TLS with SRTP

Dear all, I am seeking mobile APP and software developer to integrate ZeroTier into LinPhone softphones (Mobile APPs ANDROID & iOS & desktop applications for Windows and MAC) . both are opensource and it should be straight forward task , thanks for the available documentations for both , you may find details on The integration part: Is simply making ZeroTier embedded into LinPhone ZeroTier should work inthe backgroud upon launching, and would ask for the "Network ID" Upon adding the "Network ID" , by default it the connection estabish shall be pending till I authorize the connection "ZeroTier Address or AKA Node ID" from my end, (this connection

I am using Elastix 2.4 call center module. I'm not having any problems. I just can't install WebRCT. That's why I'm looking for an alternative solution. Things I need; A stable working call center module. WebAPI support for call center (I'm using Elastix ECCP) I want API support like this. I want WebRTC. I don't want my employees to use softphones like Xlite, Zoiper.

I want a professional looking flyer to market my telecommunications business. I want it to have 5 sections with photos and description 1. Data cabling 2. Internet Services and Onsite PBX 4. Headsets Voip Desk Telephones and Softphones 5 Camera Systems for Businesses

I want a professional looking flyer to market my telecommunications business. I want it to have 5 sections with photos and description 1. Data cabling 2. Internet Services and Onsite PBX 4. Headsets Voip Desk Telephones and Softphones 5 Camera Systems for Businesses

We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/d...chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party...

Set up Asterisk PABX and integrate with proprietary CRM. Asterisk to have web interface to manage DDi routing and ring groups. API for CRM integration will be available. An option to switch easily between desk phones and softphones

I need a Sip Client Webphone so I can integrate this into my CRM as click to call button for my clients so basically my agents will see the clients details and when they click on the call button a call will initiate to the client number via Sip Client Webphone and WebRTC and right in the CRM agent talks with the client. I don't want normal integration to other platforms and softphones like zoiper, 3cx or others I want something like: but for some reason these are not working for my sip they asking AOR or WSS which my PBX/VoIP doesn't provide so it should connect to sip only using username, password, and domain of PBX like zoiper connect.

Hello, we need a platform where we can launch our VoIP business. **** This is second attempt to get this done.. I will give you the full details of my last experience. I will give you access to what we already. If you can fix i...second attempt to get this done.. I will give you the full details of my last experience. I will give you access to what we already. If you can fix it,, then great.. If not, then we need to either build our own system.. Or use a White label Cloud PBX solution.. -We need complete billing platform. ( I don't want to walk you through what is needed in a working functional VoIP Billing platform. -Softphones tied into the billing platform.. -Easy customer onboarding Please ping me, only if you have experience in doing such a job.. And you have sample wo...