Mute fring sipemplois

...expertise in VoIP, Asterisk PBX, and Linux to configure our GoIP8 device. Our goal is to set up GoIP8 as a SIP trunk within FreePBX 14. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in configuring GoIP devices and SIP trunks. - Examples of previous projects where you successfully configured similar VoIP setups. - An outline of your approach to configuring GoIP8 as a SIP trunk within FreePBX 14. - Your proposed timeline for completing the configuration. - Your pricing structure for this configuration project. **Note:** We are looking for a reliable and efficient configuration of GoIP8 as a SIP trunk within our FreePBX system to ensure the smooth operation of ou...

une application Dialer qui peut faire un appel GSM ou SIP. Au moment de l'appel il vous demande si vous voulez passer par le GSM ou le SIP. Si vous choisissez le GSM c'est l'appel normal qui est lancé et si vous choisissez le SIP, l'appel passe par un PBX. Le SIP supporte le codec G729g. Les parties GSM et SIP sont toutes intégrées dans l'appli. Vous devez fournir le code source pour nous permettre de modifier certaines parties de l'application.

Bonjour, Je suis le fondateur d'une application poussant à la rencontre Mi. Nous sommes entrain de refaire l'application afin d'être plus attractive et fonctionnelle. J'aimerai intégrer dans cette nouvelle version de l'intelligence artificielle via un assistant virtuel qui aiderait les utilisateurs lors de leurs chat, les chats serait donc à 3 avec l'assistant ( qui pourrait être mis sur mute ) qui rebondirait sur leur conversation, les relancerait quand il y'aurait des blancs, leur donnerait des conseils... comme un véritable "assistant coach " pourrait le faire. J'aimerait également que l'assistant soit consultable en dehors des conversations pour pouvoir par exemple conseille...

Bonjour, Nous recherchons un intégrateur spécialisé VOIP capable d'installer la pile complète de logiciel Goautodial 3.3 et y installer un trunk SIP pour permettre la mise en place de campagne d'appels sortantes (sondages, prise de rendez vous). Nous fournirons un serveur dédié (idéalement OVH) qu'il faudra paramétrer intégralement et installer avec les logiciels nécessaires au bon fonctionnement de la suite GOAUTODIAL 3.3, il faudra également prévoir une migration vers Goautodial 4.0 dans les prochains mois. Possibilité d'infogérance par vos soins du serveur dédié aux campagnes d'appels.

Bonjour à tous Voici mon projet et mon besoin. Je possède : 1 téléphone IP 1 numéro de téléphone SIP fourni par OVH 1 serveur dédié sur lequel est installé KAMAILIO. Mon téléphone IP était directement connecté à ma ligne SIP chez OVH (Je pense que ma ligne SIP chez OVH est installée sur un serveur ASTERISK chez OVH) Mais depuis peu, mon FAI bloque les enregistrements et appels SIP. Mon projet est de configurer un serveur intermédiaire KAMAILIO pour l'utiliser en tant que PROXY SIP pour enregistrer mon téléphone IP sur KAMAILIO sur un port différent du 5060 et que ce dernier relais les appels en...

Bonjour, je chercher un développeur pour coder une application sous android et IOS avec les spécifications suivantes : 1- Appelle sortant via un server(SIP/IAX). 2- Forward les appelles Voip vers GSM avec un limitation de duré max 10 minutes/24 heures( Call voip ---->GSM) 3-l'application doit être exécuter sur le smartphone en arrière plan. d'autres points doivent être discuter merci de bien vouloir me contacter.

La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à - installer un trunk SIP sur un serveur Asterisk - installer minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel - installer un panel de gestion (XIVO ou autre)

La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à terminer la configuration du logiciel VOIP Xivo sur un serveur VPS Linux : installation d'un trunk SIP installation de minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel

Cherche particulier ou centre d'appel qui utilise l'offre "call center" de ou ippi fr. Pour plus de Détail. veuillez me contacter

le chat sera en flash comme123flashchat ou en html5 avec un large choix d'option...flash comme123flashchat ou en html5 avec un large choix d'option que l'on puisse mettre un templates dans chaque room de nombreuses commande pour les admins modérateurs (qu'on puisse donné les commandes qu'on veut a tel ou tel modé , une icone spécifique qui se mettra devant le pseudo dès lors de la modération, un panel admin bien conçu historique de chat privé,public les commandes de bases /mute /unmute , ban unban , banip (duree motif etc..) move(deplacer un chateur dans le salon desiré, kick (motif durée par qui pareil pour les autres commandes) warn pseudo motif (utilisateur recevant un popup sur so...

Bonjour, Je représente une PME spécialisée dans les équipements de sécurité (alarme, vidéosurveillance, contrôle d'accès, etc...) Nous devons remplacer notre bon vieux PBX PSTN par une solution IP. Nous avons opté pour le model Grandstream IP PBX UCM 6104 Nous allons utiliser six internes Grandstream GS-GXP2160 et des 3 lignes SIP. Nous cherchons une personne intéressée et capable de mettre en service notre système. Nous devons pouvoir utiliser ces services. (indicatif) - Répondeur avec horaire d'ouverture par créneau horaire - Intendant automatique avec 4 départements cibles - Mise en attente avec musique d'attente modifiable - Rép...

Nous avons un serveur asterisk configuré avec quelque sip trunk ovh. Nous avons des problemes dans la presentation des alias dans les appelles sortant. Nous avons besoins de quelqu'un qui connait bien se fournisseur Merci

Se connecter sur l'équipement sip goip, lire les statistics d'appels, et déduire ceux qui ont fait au moins x nombre d'appel échoué depuis la derniere verification. Relancer les appels sur les sims d'une durée de y minutes, avec le numero à appeler en input

Player radio windows à faire télécharger aux auditeurs et artistes avec plusieurs liens disponible dessus avec réglage du volume et avec l'option démarrage à l'ouverture de windows et mode mute .

Dans le but d'optimiser vos revenus 3D MULTIMÉDIA propose a ces free lancers de revendre ces solutions call center tels que ; interface , CRM , numéros surtaxés de tout destination ( compte sip ) , script de prospection , suivi technique , formation coaching aux centres d'appels intéressés par la réception d'appels et la prospection sur différents campagnes tels que LES JEUX CONCOURS , LA PUBLICITÉ , LA VOYANCE AUDIO-TEL .........etc Le but est de se connecter sur les forums de discutions et passé des annonces qui vous emmèneront a négocié en tant commercial avec vos clients centre d'appels de la solution création de trafic , le client disposera du pack gratuitement en ...

Bonjour, Je suis a la recherche d'une personne pour l’installation du CRM GOAUTODIAL qui est en open source bien sur : 1- L'installation se ferra sur un cloud de chez OVH France. 2- Le menu de GOAUTODIAL devra être en francais. 3- Un nom de domaine "" devra pointer dessus (L’accès a GOAUTODIA...pointer dessus (L’accès a GOAUTODIAL se ferra depuis se domaine par login et mot de passe sécurisé. 4- Conseils pour l'achat du serveur OVH. 5- Grande capacité de clients on appels simultané et compte admin et agent . 6- GOAUTODIAL devra être totalement personnalise "crutomisation" par mon logo et celui du client. 7- Payement par PAYPAL uniquement. 8- Assistance au paramétra...

As discussed.

salut Write an Android application appelle et partage fichier avec sip via wifi wifi+ appelle avec sip + partage fichier voir video qui ajouter je suis etudiant jai besoin bon prix

I'm looking for an experienced freelancer to help me set up the SIP locally to connect incoming calls to an AI script I have. This setup has to be done on VICIdial using the QuestBlue SIP. The main objective of this project is to establish a call response system. As someone with a pre-built AI script, I need you to link it to the incoming calls on the VICIdial platform. Your expertise will include, but not be limited to: - Proficiency in VICIdial: You must have previous experience working with this platform. - Familiarity with SIP set-ups: QuestBlue SIP experience would also be a bonus. - AI Integration: Able to connect incoming calls to an AI script. Please ensure these skills align with your capabilities before bidding on this project.

Construir un circuito PCB que: disponga de 2 entradas de audio stereo (Audio_In), para poder analizar y determinar continuamente, cual es el nivel máximo de volumen conjunto durante 30 segundos. Si dicho nivel máximo es SUPERIOR a un nivel establecido "manualmente" en el circuito (con un potenciómetro), entonces las salidas de audio (Audio_Out) del circuito deberían pasar a cero (Mute), en caso contrario las salidas de audio stereo deberían ser idénticas a las de entrada. (Audio_Out=Audio_In)

I'm seeking a skilled artist to conduct a 'sip and paint' workshop in a larger art exhibition context with more than 20 participants. Details: - The participants are novices in painting and hence, ideal candidate should be proficient in teaching beginners. - The key focus of the workshop will be to enage the guests, teach them few basic painting skills - oil, acrylic, landscape or fluid art. Skills Required: - Excellent painting skills - Strong teaching abilities, especially handling a large group - Prior experience in conducting workshops So, if you're someone who has experience in teaching novices the beautiful art of Landscape Painting, you could be just the one I need. Looking forward to work with a patient and engaging artist who could make this an u...

I'm looking for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom V...for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom VoIP libraries to make IVR calls. Here's a brief on what I need: - IVR Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integration: You'll need to integrate the bot with an existing SIP server. Ideal Freelancer: - Proficient in Python, especially in developing Telegram bots. - Experience with VoIP libraries, particularly in the context of IVR calls. - Familiar with SIP server integration. - Strong understanding of DTMF technology. If you're confident in your ability t...

Requiero configurar una troncal PJSIP en servidor con asterisk 18 y una troncal SIP en un servidor Asterisk 11, hay que realizar ambas configuraciones para dejar funcionando el servidor de telefonía.

We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...

...intensive research. Expertise working on softwares, call center, robocaller and conversational AI outbound calls for call center. The ideal candidate should have a proven track record in creating, managing or implementing cloud-based robo calling systems and also conversational AI calls that is good with asian language, especially Bahasa Indonesia. the subject of the research must have : telephony (SIP/VOIP) a working AI model or open to integrations with AI Basic Telco Services features such as, Campaign Calls, scheduler, call recorder) Wide system integration capability Macro/logic/script editing(for robo caller,pre recorder or AI generated TTS) white label ready or not. KPI analysis The Goal of this research is to figure which or what is the best system/platform/Integratio...

...web-based media players - Knowledge of modern UI/UX design - Previous work with live audio streaming. I want the radio player to support icecast / shoutcast and allow users to switch between multiple radio stations. Each radio station should have: * Meta info such as whats currently playing on air like images, song name, artist name, podcast name. * name of station and location * Pause, play, mute, volume, * What previously played * Whats playing next (small upcoming embed schedule) AN API will likely need to be created, we will pay extra for api to be created. An api is needed for meta whats playing from radio stream. I want the design to have header and footer. I do want back-end admin could be simple UI or a text file to control player such as adding, deleting, editing...

We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...

As an experienced tech professional, I'm seeking someone who can assist me with setting up a SIP trunk with VoIP Unlimited, and also configuring VoIP extensions for users on my existing Asterisk server. Key Requirements: - Detailed knowledge of Asterisk server - Expertise in SIP trunk setup in VoIP Unlimited - Skills in configuring VoIP extensions for users Your role will be crucial in the success of this part of the project, and will demonstrate your understanding of Asterisk servers and VoIP functionalities. A proven track record in this type of project will be advantageous.

We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...

...and detail-oriented individual who can accomplish a specific task for me. With your proficient Windows operating system skills, browsing acumen, and technology savviness, you will navigate a particular website while a screen-record captures the process. Your job will simply entail: Deliverable: Two screen recording videos (maximum of 5 minutes each) where you act as a user. Important Notes: • Mute your microphone while screen recording. • Do not reveal your identity. If you log in from your account, cut that part from the screen recording. • Do not reveal your location. Choose a random location within Australia. • Do not complete the payment on the site during checkout. Website 1: Moon Phases by WillyWeather Task 1: Current Weather Check: • Scenari...

I'm looking for a seasoned app developer with expertise in Linphone and SIP to help with customizing the open-source VoIP service. In this project, you will: - Compile Linphone with my brand logo, offering a personalized look to the application. - Integrate the app with my push server. This will involve modification of existing features to ensure a seamless flow between the app and the server. Given that the specific features to be modified weren't mentioned, I assume that you will provide the details later on or the freelancer will need to assess the necessary changes upon reviewing the current status of the app. Proficiency in VoIP/SIP and experience in both Android and Apple app development is required. Familiarity with app store submission procedures will be...

I'm seeking an expert in SIP and VoIP to assist me with various issues within my PortaOne system. Key Requirements: - Extensive knowledge of SIP and VoIP solutions, particularly with PortaOne - Proficiency in call routing and configuration for diverse scenarios - Proven experience in troubleshooting and resolving various issues - Clear communication skills with the ability to provide detailed explanations and solutions.

We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...

I have installed FreePBX - distro install. - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asteri... - My extension is registering fine. - When I call another extension, call rings but there is no audio - When I call external number, call rings but there is no audio Error message on Asterisk interface is: [2024-04-05 04:17:09] NOTICE[2335]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/1011-0000000b' for lack of audio RTP activity in 30 seconds SIP NAT is enabled Firewall is disabled SIP NAT Settings > External Address > Public IP Address is added I need someone to check this over Anydesk & ...

...automatically play on mute, and vice versa. No play or pause button should be visible; the video should play automatically when it appears on the timeline. If the video is initially played on 'mute,' and the user taps on the video to watch it as a reel for the first time, it will play with audio. However, for subsequent views, if the user mutes the video and then taps to watch it as a reel again, it will play on mute consistently. Timeline Interaction: User scrolling should stop the audio of the previous post automatically. Once a video on the timeline ends, a click on the video should allow the user to watch it again in the reels. While a video is playing on the timeline, a user can click on it to enter the reel session, and a back button must be provid...

...analyst navigating spreadsheets and avoiding office politics; the next, he's thrust into an ancient realm of magic and mayhem, summoned by a desperate sorcerer seeking a champion to save their world. But Silas? He's anything but a fighter. His battles are against deadlines, not dragons. Yet here he stands, chosen to be a guardian in a world that's as beautiful as it is deadly. With the help of a mute child, who is as deadly as she is silent they must navigate treacherous terrains while learning to master his new abilities in order to survive. True power isn't just about mastering your weapon, but also mastering yourself. With an ancient evil stirring and the fate of the realm in his unsteady hands, Silas must confront his fears, embrace his destiny, and disc...

...Jose do Rio Pardo, this coffee reflects a tradition of excellence since 2003. It's part of the Mata Atlantica project, underscoring a dedication to rainforest conservation. Indulge in a cup that celebrates both exceptional taste and ecological commitment. For the 6th image which is the Colombia photo of a street in Cartagena there's also quite a bit of text to add: Savour the luxury with every sip of our Colombian House Blend, a symphony of caramel, chocolate, and a hint of orange zest. This exquisite blend is a testament to the lush landscapes at the Andes' foothills, where our Colombian Excelso beans flourish in the nutrient-rich volcanic soil, capturing its essence to perfection. Carefully roasted to a medium perfection, our Excelso beans unveil a delicate b...

looking FreeSWITCH and ASTPP developer customise billing solution and Customer Management more details please disucss here Who can send request here 5+ years of industry experience in developing, deep knowledge PBX and Sip server SIP Development experience. Must be aware of Sip and webrtc integration. VOIP software development. Good Knowledge in PBX, SIP, RTP protocols. Worked on Queue, IVR and Voicemail related applications

...tithe underneath it but the label should only have english words on it. This logo will likely be redone over but if you include a new logo in your design that I like I will pay extra for it but the main focus is the label. The label size is 19cm x 15cm The bag is white. I need the front and back label. ~~~ Front label text: Sharpen Focus Boost Energy Strengthen Immunity Calm Stress Every Sip Supports French Vanilla 100% Organic Non GMO 30 Day Supply Net Wt 12.7oz (360G) -------------------------------------------- ~~~ Back label text: The nutritonal facts are attached. Store in a cool, dry place to preserve freshness. Brew the Perfect Cup For the ideal experience, use 1 tablespoon of coffee for every cup (8 oz) of water. Adjust to your taste. Our Mission At '...

I'm in need of a proficient specialist to establish my SIP trunk and an IPPBX server. Here are the functionalities I require: • Making both inbound and outbound VoIP calls. • Call recording, call forwarding, and voicemail to email features for my IPPBX server. You'll also need to set up three or more SIP trunks. Experience with VoIP technologies and SIP systems is crucial for the success of this project. Knowledge of REST APIs will be advantageous. Please ensure your proposal demonstrates your related past projects and expertise.

We are a vibrant cold-pressed juice bar on the lookout for an imaginative and skilled graphic designer to revamp our digital screen menus and create a detailed standing printed menu. Our brand prides itself on delivering health through every sip, offering a range of cold-pressed juices, smoothies, and health shots. We aim to capture the essence of our healthful and flavorful offerings in our menu designs, adhering to our minimalist brand aesthetic. Objectives: Digital Screen Menus: Design three clear, captivating digital screen menus that showcase our product range with prices. The design should be easily readable from a distance and engage customers with its visual appeal while following our minimalist brand guidelines. Printed Menu Design: Create a detailed printed standing menu...

We are looking fo...of SCP and SRF. Experience in handling interfaces of SCP SRF like INAP, SIP, Webservices, Websockets WebRTC (in the context of SCF and SRF) ISUP over SS7 Experience in Dialogic API for SS7 and INAP Experience in using JSS7 API from RestComm or it's variant of Mobius JSS7 or any of the varient IVR implementation with Freeswitch on SS7 Knowledge and work experience on YATE SS7 implementation. Anyone with following expereince and skills also may apply Consultant - CRBT worked in CRBT and similar products Having knowledge in the following protocols a) INAP over SIGTRAN/SIP b) SIP/RTP c) ISUP over E1 3) M3UA over SIGTRAN Knowledge on working with Freeswitch/Asterix and IVR solutions Work experience on working on E1 and SIP/RTP call scenari...

...multiple domains for me. The bot's duty would be to monitor the websites' state via three simple APIs(also checking downtime state and response time), with the ability to send different notifications as Telegram messages to the subscribed users. The frequency of alerts and other variables should be customizable. The bot should have UI menu to manage (add, delete domains, change customized variables, mute notifications for some domains or change monitoring mode) Alerts should be dispatched at specific times and also triggered by certain events. Experience with Telegram bot programming, API integration, and alert systems is essential. Key Requirements: - Monitor multiple websites' states using simple APIs - Send alerts to subscribed users (whitelisted in the in...

i need someone to teach me how to upgrade firmware of cisco 7821-k9 to make it use sip protocol to hook it up on asterisk pbx More details: Which specific features do you require for your Cisco 7821-K9 SIP protocol project? upgrade firmware , connect it to asterisk as sip extension Which version of firmware would you like to upgrade your Cisco 7821-K9 SIP protocol to? Latest firmware version What functions do you require for the SIP extension with your Asterisk system? Call Recording, Call Transfer, Multi-Line Functionality thank you very much

I require the expertise of a skilled freelancer to assist in the configuration of our Sangoma Vega 400g Voice Gateway device. It's critical that you have familiarity with E1 ISDN to SIP trunk conversion, due to our primary objective being system compatibility upgrading. Key responsibilities: - Customize settings on Sangoma Vega 400g Voice Gateway according to specific requirements. - Setting up a connection with Swedish Telco, Telia. Ideal Candidate: - Previous experience with Sangoma Vega Gateway configuration. - Knowledgeable in E1 ISDN to SIP trunk conversions. - Familiar with Telia's connectivity. - Quick and efficient problem solver, without sacrificing quality results.

I am looking for a specialist to integrate WhatsApp with SIP on Linux, focusing primarily on enabling audio communication and text messaging functionalities. The end goal is to create a seamless communication platform for efficient interaction. Key tasks include: - Designing and implementing a solution that allows WhatsApp to work in sync with SIP protocol - Ensuring the integration supports both audio calls and text messages. A deep understanding of both WhatsApp API and SIP protocol is essential, as well as experience in Linux system administration. Knowledge of VoIP would be a plus. Your delivery should be a working, robust integration that ensures smooth and seamless operative communication. Please bid with your proposed solution.

I have a Twilio account with sip trunking set up, and I've install Asterisk on Arch Linux, I've attempted to set up the config but have not been able to. I'm looking for someone to set up a basic config where I can send and receive phone calls. The details don't matter, I just want to get it working so I can adjust it once the simplest config is working. If interested please bid the amount you are able to do this for, and include the word "briefcase" in your bid so I know you've read the description and can complete the project for the amount bid.

se necesita un software que permita gestionar las llamadas voip configurable de preferencia que admita SIP Trunking, el aplicativo debe permitir gestionar las llamadas, medir el rendimiento del usuario que realiza las llamadas, debe tener tener un marcador de numero al que se va a llamar configurable(auomatico / manual / random) de preferencia el freelance debe tener un proyecto similar o terminado

I'm seeking a highly competent Investment Advisor who can guide me in the areas of Insurance, SIP and Policy. The role is focused on life insurance advice and strategies for robust legacy planning. - Expertise: You should have extensive experience and knowledge of life insurance products and policies, Systematic Investment Plan (SIP), Insurance policies. - Task: I require in-depth analysis of my current situation and guidance on where to direct my resources for optimal legacy planning. This will include assessing risks, evaluating potential returns, and designing a balanced legacy-focused investment plan. - Aim: Primarily, you will help me plan for risk cover and wealth accumulation, while particularly focusing on legacy planning via life insurance products. Plea...

I'm seeking an experienced app developer to integrate the latest PortSIP SDK into a richly featured communication app for both iOS and Android. This app will connect with my existing Asterisk server through APIs. Key Features Include: - User creation - Real-time balance display - Call-making functionality - Fully integrated payment gateway - Text messaging - SIP voice calls (Not video calls, just normal SIP calls) Necessary Skills and Experience: - Proficient in iOS and Android app development - Proven experience with PortSIP SDK - Familiarity with Asterisk and its relevant APIs - Skills in developing chat features, specifically text messaging and voice calls, within an app - Experience in implementing a payment gateway in an app Please note, I have access to and can ...