Sip dialer g723trabajos

Aplicación para Android para llamadas de teléfono seguras, con RSA y AES, punto a punto, VOIP,SIP, GSM, CDMA, 2G, 3G.

Necesito un sip softphone descargable para android y apple. Con el logotipo de mi web y el nombre de mi web. Para que mis clientes lo puedan descargar de mi web. Espero prupuestas serias y confiables. Gracias.

Estamos buscando a un telemarketer que pueda trabajar desde casa para nutrir y convertir a los leads que generamos desde diferentes canales.  Referente al coste de llamadas, se facilitará un usuario SIP de telefonia para gestionar sus leads, tanto en entradas entrantes (si asi lo considera) como en la emisión de las llamadas a dichos clientes.  Preferiblemente Español, para realizar llamadas en territorio local.  ?en breve realizaremos contrataciones en Argentina para antender territorio latam Responsabilidades: - Colaborar con los equipos de ventas para identificar oportunidades de negocio - Feedback - Gestionar la salida de llamadas a clientes actuales o potenciales clientes. - Entregar preparados las conversaciones de ventas y la lectura de...

Necesito que me proporcionen un softphone sip para android, pc y apple, el cual yo pueda configurar y personalizar con mi nombre comercial y mi logo comercial. No necesito que me hagan el trabajo, lo que necesito es que me den la posibilidad de hacerlo yo mismo. Pago por que me den el acceso a hacerlo. Espero respuestas. Gracias.

...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 pequeña compa&nti...

Nuestro profesional programador y/o desarrollador debe llevar acabo la virtualización un servicio de telefonía IP (PBX Virtual) en un entorno Virtual (Amazon Web Service), implementando software open source que mas aporte a la plataforma de comunicaciones a desarrollar (Asterisk. Elastic. FreePBX, etc.) integracion proveedor troncales SIP (DID), El PBX VIRTUAL se requiere para la compañía y para nuestros clientes, es decir que la plataforma debe contar con portal para clientes y seguridad.

Customizacion de una aplicacion android existente. La aplicacion ya esta desarrollada, solo hay que personalizarla para usar en nuestra empresa. Es una aplicacion del tipo SIP Client para Android del tipo Bria, Zoiper, Sipdroid, etc. Con un tiempo de trabajo de 30 horas aproximadamente.

Necesitamos para un proyecto complementario de un cliente, una persona o empresa, que pueda realizar la migración de los siguientes modelos de teléfonos CISCO a versión SIP y registrarlos en una centralita Asterisk (open source) : 7942 7936 7970 7940 7912 7911 Se necesita realizar tanto la migración, con su testeo de funcionamiento y registro en la centralita Asterisk correcto, así como proveer a la finalización de un manual de la migración y funcionamiento del sistema.

Necesitamos una aplicación que interactue contra un webservice que tiene tres métodos; provisión, saldo y llamada. Debe tener cuatro actividades; agenda, favoritos, histórico, y dialer. Hay alguna funcionalidad más que necesitamos incluir; - El usuario recibe un SMS con el código de activación y la aplicación debe poder leerlo automáticamente. - Cuando fuera de la aplicación el usuario pulsa o escribe un numero de teléfono, debe preguntar si quiere hacerlo por nuestra aplicación. - Cuando invocas el método de llamada del webservice al cabo de unos segundos recibes una llamada que hay que descolgar automáticamente. El código fuente y la propiedad intelectual será nuestra...

necesito modificar un cliente SIP y agregarle que descargue la libreta de direcciones desde un ftp, mi presupuesto es 40 usd, el cliente sip solo abria que editarle un poco la estica y seria agregarle la parte de ftp

Hola Necesito configurar un Elastix que ya tengo instalado, instalar chan-sccp y convertir 2 cisco 7961G que estan en SIP, configurar bien el firewall para seguridad

Necesitamos una aplicación que interactue contra un webservice que tiene tres métodos; provisión, saldo y llamada. Debe tener cuatro actividades; agenda, favoritos, histórico, y dialer. Hay alguna funcionalidad más que necesitamos incluir; - El usuario recibe un SMS con el código de activación y la aplicación debe poder leerlo automáticamente. - Cuando fuera de la aplicación el usuario pulsa o escribe un numero de teléfono, debe preguntar si quiere hacerlo por nuestra aplicación. - Cuando invocas el método de llamada del webservice al cabo de unos segundos recibes una llamada que hay que descolgar automáticamente. El código fuente y la propiedad intelectual ser&aac...

Estamos buscando un Desarrollador CSS / HTML5 que pueda integrar un webphone en nuestra web corporativa. En concreto queremos que se integre SIPML5  El webphone deberá cumplir con lo siguiente: 1) Diseño acorde a nuestra imagen corporativa 2) Logueo automático de credenciales SIP en base a la autenticación del usuario en nuestra web. 3) Predefinir configuración del webphone contra nuestro SIP gateway 4) Incorporación del webhone en nuestra web. Se creará una sección nueva para incluir el webhone, nuestra web está programada bajo Symfony 2.1 por lo que se valorará experiencia en este enterno para incorporar el webphone en nuestra web.  

...tenemos un precio bajo y mucho que ofrecer para el cliente, es importante que los vendedores conozcan la importancia del posicionamiento y la presencia web para los negocios el costo de nuestro pack es de $250 y ofrecemos $75 por venta ademas de un residual despues de 90 dias el trabajo es cerrar ventas por telefono con negocios de USA por eso requerimos buen ingles para este trabajo ofrecemos dialer y lista de contactos o leads saludos...

Configurar remotamente un CP-8961 con parametros de un proveedor SIP en internet, (no usa call manager). Se facilita ingreso a un equipo en la misma red lan para configuracion del dispositivo, El servidor SIP esta publicado en internet.

...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 pequeña compa&ntild...

Tengo un equipo gateway GSM a SIP marca "Portech mv370", el cual trabaja con una central telefonica IP. Dicho equipo posee 1 linea GSM. El problema que tengo es que algunas su linea no se registran correctamente contra la telco. Lo que necesito es cambiar el GSM s/n (o imei) de su modulo gsm. <span style="font-size: 13px; line-height: 1.6em;">Puedo proveer acceso telnet y web para evaular si esta en condiciones de realizar este proyecto.</span> gracias

sip softphone android para asterisk, mostrar el saldo, video llamadas, crear usuario automaticamente y registrarse

Queremos una página web que tendrá opcion de crear cuentas SIP y gestionarlas, adquirir nuestros servicios y pago seguro. Nos interesa que se nos diseñe la identidad corporativa y también que el proyecto no sea solo la creación sino también colaboración a largo plazo.

Tengo un equipo gateway GSM a SIP marca "Portech mv378", el cual trabaja con una central telefonica IP. Dicho equipo posee 8 lineas GSM. El problema que tengo es que algunas de sus lineas no se registran correctamente contra la telco. Lo que necesito es cambiar el GSM s/n (o imei) de algunos de sus modulos gsm. puedo proveer acceso telnet y web. gracias  

I need an application to change the charging system elastix. I want you to see in real time (online). call a client or difficult to appraise the call and the value of that call. change the end customer interface, to make it l...aplicaciones para telefonos inteligentes, androides, iphone  o utilizar estas caracteristicas para algo nuevo basado en asterisk   Especificaciones: softswitch: asterisk lenguaje de programacion: php, preferiblemente pyton base de datos mysql    o    postgresql SO: GNU/Linux Debian       Técnicians features Soporte SIP, H323, IAX, Dahdi, E1/T1, entre otros Codecs G711, G723, G729, G726, GSM, chan_skype Desarrolado sobre Asterisk y Linux Debian - Super Estable!! Interfaz Web Basada ...

Requerimiento técnico de plataforma de audio conferencias bajo asterisk/elastix para alojarlo en la nubey contratar un servidor virtual La plataforma sea Asterisk Soporte hasta 200 llamadas concurrentes de el proveedor de telefonía que actualmente me entrega desde un softswitch con protocolo sip (el ancho de banda ya esta contemplado para este tipo de plataforma) Se requiere que sea compatible con los codecs G.711 y G.729 Soporte uso de Softphone Debe poder ser Configurado en modo gráfico (Elastix o alguna otra) Numero de salas debe ser estático (nunca expire el código de la sala) y dinámico(que se puedan crear al vuelo) generar código de organizador y código de participante por cada sala que se a dada de alta en el sis...

I require an experienced and proficient developer who is familiar with both WebRTC and Asterisk. The objective is to expand the functionality of my current Asterisk setup by integrating it with WebRTC. The specific features to be integrated are: -Voice Calling -Vi...Asterisk. The objective is to expand the functionality of my current Asterisk setup by integrating it with WebRTC. The specific features to be integrated are: -Voice Calling -Video Conferencing This project seeks to enhance the communication capabilities of my Asterisk setup, thus the ideal candidate would have considerable experience managing such integrations in the past. Sound knowledge of VoIP, SIP, and related technologies will be greatly appreciated. An understanding of the Asterisk framework and its API will be...

...Border Controller (ASBCE) - specifically, to set up and integrate it with SIP Trunk Provider lines and Avaya IP Office Server Edition. AVAYA IP Office servers are already configured and calls working internally. - The tasks involved are both the installation of the ASBCE and the setup and integration with the SIP Trunk Provider. This requires a comprehensive understanding of both Avaya systems and SIP technology. - The primary goal for this configuration is to achieve a seamless integration between the ASBCE, the SIP Trunk Provider, and our Avaya servers. The focus is on ensuring robust communication and operation between these systems. - Basic training on how to restrict external call access to everyone, give dedicated SIP line to one user etc. Th...

Are you a skilled communicator with a knack for scheduling appointments? Glimmer Grove Design is seeking freelance appointment setters to assist us in reaching potential clients and scheduling Zoom calls to showcase our beautifully crafted websites. Responsibilities: - Utilize provided dialer, script, and lead list to make outbound calls to businesses without websites. - Engage with business owners or decision-makers to discuss our website offerings. - Schedule Zoom calls to present our pre-built website demos. - Maintain accurate records of appointments scheduled. - Requirements: - Excellent communication skills, both verbal and written. - Comfortable making outbound calls and engaging with potential clients. - Ability to effectively follow a provided script while adapting to co...

I'm looking for a professional...me set up Audiocodes with Microsoft Azure Direct Route to integrate with a PBX system. Key Requirements: - Extensive experience with configuring Audiocodes for call routing - Proven track record in setting up SIP trunking - In-depth understanding of Microsoft Azure, particularly with Direct Route - Ability to seamlessly integrate Audiocodes with a PBX system The primary goal of this project is to ensure that the Audiocodes system works efficiently and effectively with the Microsoft Azure Direct Route, specifically for integration with a PBX system. You should have a solid grasp of SIP trunking and the ability to deliver a setup that guarantees optimal call routing within the integrated system. Note : i need with documents how to do so...

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

https://www.freelancer.com/projects/voip/Kamailio-SIP-Proxy-RTPengine/details

I'm seeking a professional who can configure my Kamailio SIP proxy and RTPengine. The main purpose of this project is to set up call routing and load balancing. Key Responsibilities: - Configure Kamailio SIP proxy for call routing - Implement load balancing features - Ensure compatibility with SIP over UDP Ideal Skills: - Proficient with Kamailio SIP proxy and RTPengine - Strong understanding of call routing and load balancing - Familiarity with SIP over UDP I do not require integration with any specific external systems or services. The primary goal is to optimize my setup for efficient call routing and load balancing. Please share your relevant experience when applying. It should read from a text file, xml or database the list of servers it nee...

...you per qualified appointment you set. --- We have a high demand for at least 10~18 appointments weekly, and we have multiple areas we can assign you based on your results. So, you can easily earn $750 weekly if you provide quality appointments. --- We are looking for a long-term partnership. We have demand for multiple locations in the US, Canada, and Australia. --- You need to have your own dialer and leads (We have ppl working with us using Google Maps and --- Qualification criteria --- A business with the REAL possibility of becoming our client: NOT pushing appointments, NOT people just looking to compare prices, NOT people happy with their current services. They need to provide a reason on the call why they want the appointment, for example: "I'm not satisfi...

Are you a skilled communicator with a knack for scheduling appointments? Glimmer Grove Design is seeking freelance appointment setters to assist us in reaching potential clients and scheduling Zoom calls to showcase our beautifully crafted websites. Responsibilities: - Utilize provided dialer, script, and lead list to make outbound calls to businesses without websites. - Engage with business owners or decision-makers to discuss our website offerings. - Schedule Zoom calls to present our pre-built website demos. - Maintain accurate records of appointments scheduled. - Requirements: - Excellent communication skills, both verbal and written. - Comfortable making outbound calls and engaging with potential clients. - Ability to effectively follow a provided script while adapting to co...

Objective Develop and integrate a SIP telephony client within the Milestone video recording software as a plugin using the Milestone SDK. This SIP client will allow the configuration of connection data with the PBX and enable the functionality to automatically answer calls so that they are recorded by Milestone. Project Scope Development of the SIP Client: Implement a SIP client that can be integrated as a plugin into the Milestone software using the provided SDK. Ensure the compatibility of the SIP client with current and future versions of the Milestone software. SIP Client Configuration: Add a user interface in Milestone to configure the SIP client. Configuration parameters should include: SIP server address. Connection port. Us...

Description: We are seeking motivated individuals to join our team as Outbound Dialers for Health Plan Eligibility. In this role, you will be responsible for reaching out to low-income prospects using the data provided, to determine their eligibility for $0 health plans. Responsibilities: Conduct outbound calls to individuals and families identified as potential candidates for $0 health plans. Engage with prospects in a friendly and professional manner to gather necessary information. Assess eligibility criteria based on provided guidelines and accurately record findings. Provide clear and concise information about available health plan options and benefits. Document all interactions and maintain detailed records of each prospect's status and outcomes. Requirements: Excellent...

I am in need of a highly capable and experienced iOS developer with demonstrated experience with the Linphone app. Key responsibilities: - You will be responsible for implementing major modifications to the existing Linphone app. - The focus will be to enhance communication services, so familiarity with VoIP and SIP protocols is necessary. - As I have a set deadline, timely completion is critical. Ideal candidate skills: - Proficiency in iOS development - In-depth knowledge of the Linphone app - Experience in enhancing communication services on VoIP apps - Demonstrable ability to deliver on tight deadlines Your experience in modifying the Linphone app, enhancing communication services and delivering within tight deadlines will be a great asset to this project.

I am searching for an efficient freelancer, who is proficient in spoken Czech/Slovakian. Your primary task will be to confirm the shipping addresses for new orders. These orders from our perfume webshop will be generated multiple times a day, so exceptional organizational skills a...Fluency in Czech/Slovakian - Excellent verbal communication skills - Customer service experience - Ability to work during specified hours - Strong organizational skills. Expect to interact extensively with our new customers, ensuring their orders are delivered seamlessly. Your affable communication will amplify their shopping experience, enhancing our brand's relationship with them. We provide dialer and everything. At the moment it's cca 30 minutes in the mornings, and 30 minutes late afte...

I need the most simple Softphone/SIP app with my logo to use or another reseller voip/phone number provider so we can make calls in/out. I need this for Android, IOS, and PC/Mac. Simple app to call in/out. I need a custom softphone app (cross-platform) developed urgently. The app should be a simple dialer for both incoming and outgoing calls, and should be connected to servers. It's important that the app displays our company logo prominently. Key Requirements: - Development of a cross-platform softphone app - Integration with servers - Basic dialer functionality for both incoming and outgoing calls - Display of our company logo Ideal Skills: - Experience in softphone app development - Proficiency in cross-platform development - Ability to integrate with

I'm in need of a SIP and Asterisk expert to help troubleshoot issues with a SIP interconnect between two companies, focusing purely on voice communication. Key responsibilities include: - Identifying issues within the existing setup - Repairing and configuring the SIP and Asterisk system Ideal skills and experience include: - Proven experience with SIP and Asterisk systems - Strong troubleshooting abilities - Understanding of VoIP technologies - Excellent communication skills

...will give you: 1. Virtual Numbers 2. SIP Trunk server I am in need of an expert who can develop a SIP Trunk server with the capability of enabling live voice stream access. Key Requirements: - SIP Trunk Server: The primary purpose of this project is the development of a SIP Trunk server. It is essential that the chosen freelancer has experience in developing such systems. - Live Voice Stream Access: The server must offer functionality to enable live voice stream access. This feature must be robust, reliable, and seamlessly integrated into the server. - Access Control: I need the live voice streams to be secured with user-based permissions. This ensures that only authorized users can access the streams. Integration: - The primary platform that the SIP...

I am looking for an Asterisk/SIP expert to help me configure a SIP trunk. Specific tasks include: - Configuring the SIP trunk on my Asterisk server - Assisting with troubleshooting any issues that arise during the setup process Requirements: - Expertise in Asterisk and SIP configuration - Experience with setting up SIP trunks - Familiarity with troubleshooting Asterisk setups Additional information: - I already have a SIP provider - Ongoing maintenance assistance will be required after the SIP trunk has been configured.

We have a voip app, voice calling with sip and fcm setup for the push notifications, we like to have a call alert like whatsapp calling alert that can handle both foreground and background states in Android and iOS. Also mention if somebody already worked on this type of feature, bid and connect to show a demo. Feature Required: Call Notification Incoming Screen Alert on App background and foreground State both killed or behind other apps. When you accept the call redirect to the ongoing call screen and call get connected with sip.

...from ongoing voice communication. Upon recognition of specified words, the code will need to send DTMF commands via a SIM7600 or SIP/VOIP system. In order to accomplish this project's requirements: - Proficiency in Python 3.5 is a must, as the script will need to be written fully in this version of Python. - Familiarity with serial communication is key, as this will be the method through which the script interfaces with the SIM7600 or SIP/VOIP system. - A thorough understanding of Google Cloud Speech-to-Text API is essential given its integral role in detecting specified words from speech in real-time. - Experience with telecommunication platforms, particularly SIP/VOIP systems and SIM7600 modules, will be highly beneficial for successful completion of this ...

I'm currently in need of a tech-savvy individual to set up the VLAN on my managed TP-Link switch for configuring the SIP line from Tata. Key responsibilities include: - Setting up the VLAN on the switch . - The task will be performed virtually; access will be given using Anydesk. Ideal freelancer should: - Have extensive knowledge of managing TP-Link switch for VLAN configuration - Be proficient in understanding networking concepts related to VLANs

...candidate will be fluent in both English and Spanish, with a clean accent. This role involves setting up qualified appointments for our solar program through cold calling. Responsibilities: • Conduct outbound cold calls to prospects in English and Spanish, promoting our solar program • Set up a minimum of 3 qualified appointments daily during an 8-hour shift (Eastern Time) • Utilize provided calling Dialer and data to effectively reach prospects • Earn incentives based on the number of qualified appointments set up, with additional bonuses on installs • Follow up with customers to confirm appointments and provide necessary information • Update CRM system with relevant customer details and appointment schedules • Minimum 3 years of experience ...

Urgent Loocking one program for change voice in live call / Need this program working on windows for softphone (zoiper ,xlite , eyes beam ) Its will be for call center ! Sip Agent need to change his her voice immediatly and voice must be natural . Girl to man /man to girl Im pay direct 30 usd who find me this program . And propose direct on test .

...efficiently. - The endpoint devices that will connect to the Asterisk PBX are Mobile Operator issued SIP trunks. Ideal Skills and Experience: - Prior experience in setting up and configuring Asterisk PBX systems is essential. - Proficiency in handling and routing calls effectively. - Knowledge of SIP trunks and mobile operators' systems would be a plus. Specific Requirements Requirement: A simple Asterisk PBX installed on our server. It is a voice transit facility with incoming and outgoing SIP trunks. All incoming caller ID’s will be replaced by the DIDs provided for each MNO trunk. Connections 1) Multiple incoming customer Accounts from Soft switches like VOS3000 2) Multiple Outgoing SIP trunks provided by local MNO’s DID assignment/Man...

Hi Mohammed S., We have been in touch regarding PBX some time ago. I need a simple PBX installed that can recieve calls from VOS3000 and route them to SIP trunk provided by operator. The incoming caller I D to be modified to match DIDs provided by the SIP trunk. Also ability to defibe

We have a VoIP app in flutter, voice calling with sip and FCM setup for the push notifications, we like to have a call alert like WhatsApp calling alert that can handle both foreground and background states in Android and iOS. If somebody already worked on this type of feature, bid and connect to show a demo.

I'm looking for someone experienced with Asterisk to help me set up a SIP server for educational and testing purposes. The SIP server will be used with a Jio SIP trunk. Key requirements: - Configure Asterisk as a SIP server on the operating system of your choice - Set up a Jio SIP trunk - Create a demonstration of simple dialing using an open source SIP client Ideal skills and experience for this project: - Proficient in Asterisk server configuration - Experience with setting up SIP trunks - Strong knowledge of open source SIP clients - Good communication skills to help guide me through the setup and demonstration process. My budget is not very high ..