Ns2 sip Jobs

Installation eines Jitsi Meet Servers auf bereitgestelltem Ubuntu Server. Zugangsdaten stehen bereit Grundkonfiguration Jitsi für Browser und App-Nutzung Nach Rücksprache: - erstellen eines SIP Accounts - Setup FreePBX mit IVR Einwahl

Installation and maintenance of an Asterisk telephone system. Takeover of the old Asterisk system. (The old system is 15 years old). Many...etc. Provision of the data e.g. on an ASPX interface for other systems. We would like a contact in German. --- Installation und Pflege eines Asterisk Telefonsystemtems. Übernahme des alten Asterisk Systems. (Das alte System ist 15 Jahre alt). Es müssen viele Wählpläne neu geschrieben werden. Es sollen neuen Funktionen hinzukommen z.B. Chaträume für Konfenzenzen etc.. Neue SIP Telefone installieren. Intergration eines SIP Anbieters wie SipGate sowie weitere Verwendung des ISDN Adapter. Bereitstellung der Daten z.B. auf einer ASPX Schnittstelle für andere Systeme. Wir wünschen uns einen Kontakt ...

english >--------------------------------------------------------------------------------------------------------------- -- SIP client as APP for Nextcloud with JSSIP library. The following functions should be supported: To dial a keypad 0-9 *,# Answer and hang up with DTMF transmission. 1. register extension 2. make call 3. accept call german >--------------------------------------------------------------------------------------------------------------- SIP Client als APP für Nextcloud mit JSSIP library Folgende Funktionen sollen unterstützt werden: Zum Wählen ein Tastenfeld 0-9 *,# Annehmen und Auflegen mit DTMF Übertragung 1. Nebenstelle registrieren 2. Anruf ausführen 3. Anruf annehmen

Wir möchten SIP Panel Holzhaus Leichtbau-Konstruktionen in Deutschland vermarkten.

Istzustand: 3CX Telefonanlage auf einen firmeneigenen Server. (läuft bereits) ca. 5 Snom Telefone (IP per DHCP jedoch statisch fixiert), teilweise über VPN von extern angebunden. SIP-Trunk von Sipgate. Ich möchte gerne eine einzelne HTML Seite programmieren auf der der Status von ca. 5-10 SNOM Telefonen angezeigt wird. Unter Status meine ich den Zustand (Frei, Besetzt, DND, ...) Vewendet werden folgende Telefone: SNOM 760 und SNOM D735 Diese HTML Seite sollte auf einen eigenen vorhandenen Linux-Apache-Server laufen. Die Seite sollte von allen Mitarbeitern im internen Netzwerk zugänglich sein. Zugriffsgegelung ist ncht notwendig. Jeder im internen Netzwerk darf auf diese Seite zugreifen. Auch möchte ich gerne, dass der Zustand DND auf dieser Seite für...

Ich habe einen MacOS Catalina und benötige Hylafax um Serienfaxe zu versenden. Auf dem Mac soll das Paket T38Modem kompiliert werden unter root per SSH Zugang für Sie remote, bzw. sollten Sie per VNP oder Teamviewer zugreifen und darauf arbeiten, bzw. sich auskennen mit Hylafax und T38Modem. Ich versende die Faxe mit einem VoIP Account bei SIPGATE und SIP über einen SIP Server, sowie mit der command-line per sendfax 012344535234 beispielsweise dann, wie unter Linux.

Ich würde gerne eine App erstellen lassen welche also SIP Account dient - also mit der man telefonieren kann. Ziel soll es sein dass die App sich über einen QR Code selber konfiguriert mit den notwendigen SIP Einstellungen.

Ich suche jemanden, der mir kurzfristig einen Jitsi Server für Video-Konferenzen mit meinen Kunden und Mitarbeitern aufsetzt. Den Account bei Hetzner o.ä. habe ich noch nicht gebucht, hier lasse ich mich gerne beraten. Weitere Aufgaben: - Footer auf der Startseite mit Link auf unsere Website und Impressum ergänzen. - ggf. SIP-Gateway installieren (Jigasi).

Hi debjeetfailure6, I noticed your profile and would like to offer you my project. We can discuss any details over chat. Wir suchen nach jemandem, der uns bei unserer bestehenden Jitsi Konfiguration noch eine Einwahl via SIP ermöglicht. PIN Eingabe via Telefon -> Übergabe Conference ID und Join in den jeweiligen Raum. VPS mit Jitsi-Meet ist vorhanden, ein Sipgate Basic Account für die Einwahl auch. Dieser müsste zum Testen reichen, falls danach was anderes notwendig ist, könnte auch ein easybell Trunk gemietet werden etc. Was wäre der Zeitaufwand dafür in etwa das IVR zu generieren etc.?

...analytische Fähigkeiten Schnelle Auffassungsgabe und die Bereitschaft, neue Produkte und Technologien zu lernen Teamplayer mit hohem Einsatz und Engagement Deutscher Muttersprachler mit sehr guten Englischkenntnissen in Wort und Schrift Seien Sie bitte bereit, ein erstes technisches Gespräch auf dem Grundverständnis der folgenden Konzepte zu führen: DHCP, DNS, NTP, TCP/UDP, NAT, firewall, switch, VoIP (SIP, RTP). WAS WIR IHNEN BIETEN Attraktives Gehaltspaket mit Weiterbildung- und Entwicklungsmöglichkeiten Internationales Team Smart Working (2 Tage in der Woche) [Removed by Freelancer.com Admin]...

Ich suche jemanden der mir eine funktionierende Verbindung zwischen einem Freepbx und einem Digium 1-Port T1/E1/SIP Gateway (G100) herstellen kann

Configure A2Billing + Elastix 2.5 (Using DAHDI & SIP Trunk Provider)

Ich benötige einen Spezialisten im Bereich Telefonie im PSTN Netz. Die Aufgabenstellung lautet, eine Verbindung zu einem Gesprächspartner in 500 ms - 800 ms herzustellen ( es zählt die Zeit bis die Leitung besetzt ist ). Die Gegenstelle ist in 90 % der Fälle ein GSM Teilnehmer. Wie die Verbindung Aufgebaut wird ( z.B. Asterisk mit SIP Gateway oder GSM Dongles ) wird nicht vorgegeben solange das Ziel erreicht wird. Die Aufgabe ist erfüllt wenn ein Nachweis erbracht ist, dass in 100 Bespielanrufen ( verteilt auf alle Mobilfunkanbieter O2, D1 und Vodafone ) das ziel im Durchschnitt erreicht wurde. Es handelt sich hier um eine hoch komplexe Aufgabenstellung, wenn Ihr nicht genau wisst was zu tun ist oder mir einfach eine SIPGATE Anbindung verkaufen w...

...Produkten. Für unsere Kunde suchen wir eine/n SIP/WebRTC Freelancer. Ihr Profil: Erfahrung mit WebRTC. Sehr gute Verständnis für TCP/IP und SIP/RTP Protokollen. Informatikstudium oder informationstechnische Berufsausbildung. Mindestens drei Jahre einschlägige Berufserfahrung im Bereich der Softwareentwicklung. Fundierte Kenntnisse in der Softwareentwicklung mit Java, C/C++ und Skriptsprachen. Erfahrung mit der Arbeit im Linux-Umfeld Selbstständige und verantwortungsvolle Arbeitsweise, Ausgeprägte Teamfähigkeit und Kommunikationsstärke, Lernbereitschaft. Sehr gute Englischkentnisse in Wort und Schrift. Von Vorteil: Gute bis Sehr gute Kentnisse in VOIP , B2BUA und PBX (Asterisk, FreeSWITCH), SIP proxies(...

...vermittelt. Dieser kann, entweder über vorprogrammierter Schleifen oder noch besser per Schnittstelle (Befehle unseres System an den Telefonserver), eine Reihe von Personen auf Festnetz/Mobil anrufen (kein SIP-Telefon und Ausland). Beim annehmen des Anrufst wird eine kurze Bandansage abgespielt. Bleibt er in der Leitung (quasi als Bestätigung, dass er Zeit hat) wird er im First-Come-First-Serve Verfahren mit dem Kunden verbunden. Der PM soll eine optionale Moderator-Funktion erhalten. (Es handelt sich im übrigen nicht um ein Erotik-Angebot) Das Projekt soll gerne bei einem externen SIP/VoIP-Anbieter liegen, daher suchen wir einen Projekt-Planer der Erfahrung mitbringt und unsere Vorstellungen mit selbstkonfiguriertem Asterisk-System oder in Kooperation mi...

Wir haben einen lokalen PBX (Asterisk mit Elastix) im Netzwerk, der über eine IP erreichbar ist. Auf XOS und Windows benutzen wir Zoiper und x-lite um mit dem PBX zu kommunizieren. Leider gibt es für Chrome OS noch keinen SIP client. Diesen müssen wir entwickeln. Zu beginn reicht eine einfach Erweiterung, in die ich die Telefonnummer eingebe und die dann den Anruf macht. Später kann man die Funktionen erweitern. Ich weiß, dass WebRTC, HTML5 und JSSIP zur Verfügung stehen, habe aber keine Erfahrung damit. So wie ich es verstanden habe, hängt es auch stark von unserem Asterisk ab, ob es funktionieren wird. Deshalb sollten wir vorab vielleicht ein paar Tests machen.

Ich benötige GoAutodial bzw. Vicidial auf einem Server fertig Installiert haben. Es soll ein SIP-Trunk angeschlossen werden und eine API zu ZohoCRM gebaut werden. Für weitere Fragen stehe ich gern zur Verfügung.

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 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

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 ..

I have Patton Smartnode which gets SIP line in (works) and needs to output to its 4 BRI ports to connect to classic PBX (Avaya IP Office 500). SIP line works BRI Configuration to connect Patton and PBX not, this is where i need help for.

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 to help me set up a SIP trunk on my FreePBX phone system. Key Requirements: - Configuration of SIP trunking: I need you to assist in the setup of a SIP trunk on my FreePBX system. Your expertise in this area is crucial. - Connecting to a specific SIP trunk provider: I have a specific SIP trunk provider in mind, and I need you to ensure that the FreePBX system is correctly connected to this provider. Ideal Skills and Experience: - Proven experience with FreePBX phone systems and SIP trunking: I'm looking for someone who has successfully configured SIP trunking on FreePBX systems before. Please provide examples of similar work. - Familiarity with various SIP trunking providers: Your knowledge of di...

I am in urgent need of an Asterisk developer who can work efficiently on debugging and troubleshooting. Key Tasks: - Fix call dropouts - Resolve audio quality issues - Correct failed call routing The ideal candidate for this project should have: - Strong knowledge of SIP protocols - Experience with Asterisk dial plans - Familiarity with debugging tools and logs Only developers well-versed in these areas need to apply. Achieving solid results in a timely manner is of the essence.

The objective of this project involves the implementation and modification of an open SIP protocol to enable data synchronization between mobile devices and computers. The successful freelancer should possess: - A comprehensive understanding and experience with SIP protocols - Ability to implement and customize existing protocols for specific needs - Proficient in facilitating cross-device connections, specifically between mobile devices and computers - Familiarity with data synchronization methods and applications By the end of this project, we aim to have seamless data synchronization between the devices in question. As such, creative problem-solving skills and a keen attention to detail will be invaluable.

I am setting up a call center and need assistance activating SIP via VPN on Mikrotik OS. The SIP server for the call center will be accessed through a site-to-site VPN using IPsec. Key requirements include: - Configuring Mikrotik OS for SIP calls - Establishing secure site-to-site VPN using IPsec - Ensuring the SIP server is accessible through the VPN - Preparing the system to handle more than 50 concurrent calls at peak times Experience and Skills: - Expertise in Mikrotik OS - Proficiency in SIP server configuration - Strong understanding of VPNs and IPsec - Prior experience with call center setups would be advantageous - Ability to design systems that can handle high call volumes is essential Please include any relevant experience or examples of sim...

I am looking for an experienced professional to install a SIP trunk server on a Linux system. As part of the project, you should also set up access to the live voice stream with the primary purpose of real-time transcription. Key Requirements: - Install and configure a SIP trunk server on a Linux system. - Enable access to a live voice stream for real-time transcription. Ideal Candidate: - Proficient in Linux server administration. - Experienced in SIP trunk server installation. - Familiar with real-time transcription processes. - Skilled in setting up live voice stream access. - Strong problem-solving and troubleshooting abilities.

General Event Assistants wanted for a fun, science-themed event. The Lit Lab (@litlablondon) is the UK’s first and only ‘Science and Sip’ experience. In this exclusive, immersive pop-up event, guests will be transformed into scientists to create exciting drinks, conduct science experiments, and compete in challenges against other scientists. Responsibilities: - Act as general event staff for monthly events and some ad-hoc events in London. - Event dates and times TBC. These will be shared at least 1 month in advance where possible, at which time you can accept or decline within 1 week. - Temporary / day work: 4-5 hours work. This is a pop-up event that changes locations. - Setup and clean-up: Moving and arranging chairs, tables, equipment and storage boxes e...

...branding guidelines. Test modifications thoroughly to ensure stability and functionality. Collaborate with our internal team to gather requirements and feedback. Provide documentation and support for the customized software. Requirements: Strong proficiency in Visual C++. Experience with software customization, particularly with open-source projects. Familiarity with VoIP technologies and SIP protocols is a plus. Excellent problem-solving skills and attention to detail. Ability to work independently and manage time effectively. Project Details: This is a freelance, remote position. Duration: Approximately 1-3 months, depending on the project scope and requirements. Payment will be negotiated based on experience and the specifics of the agreed-upon deliverables. ...

I need an experienced professional to help fix an urgent problem in my Asterisk system that utilizes both Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP). My issue revolves around tenant to tenant recording or IVR. Specifically, all audio plays for only 2 seconds before abruptly ending calls. - Skills and Experience You should have extensive experience with Asterisk, SIP, and RTP. A deep understanding of their inner workings and potential pitfalls is necessary to effectively troubleshoot and resolve the current issue. A track record for quick and efficient problem solving is essential. Deliverables: - Diagnose the issue causing the audio and calls to end after 2 seconds - Implement a reliable solution to fix the issue Note that this project deman...

I am looking for an expert who can help me set up a SIP trunk server using Twilio as the SIP provider. This server will be used for both voice calling and SMS messaging. Key Responsibilities: - Establishing a SIP trunk server with configurations optimized for Twilio - Implementing call management features like Call Recording, Call Forwarding, and Interactive Voice Response (IVR) - Ensuring seamless integration of both voice and SMS capabilities - Providing recommendations for server security and performance optimization Ideal Candidate: - Extensive experience with SIP trunking, particularly with Twilio - Proficiency in configuring and customizing SIP trunk servers - Proven track record in implementing call management features such as Call Recording, Ca...

...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP t...

...ideally have experience creating SIP trunking solutions focused on handling incoming and outgoing calls with a Python bot. Key Features: - Set up SIP trunking for call recording and monitoring, IVR system, and SIP trunking. - Convert my analog phone system to a VoIP system. - Implement Direct Inward Dialing(DID) support for calls between multiple company sites. - Configure the Python bot to manage incoming and outgoing calls. Bot Functionality: - Answer and route incoming calls to appropriate destinations. - Initiate outgoing calls based on specific triggers or events. - Handle real-time call transfers and call forwarding. Skills Required: - Proficient in VoIP and PJSIP development using Asterisk. - Strong Python programming skills. - Experience with SIP t...

Seeking an experienced Asterisk developer to optimize the call quality of our existing VoIP system within our business operations. The ideal candidate must: - Possess 3+ years of experience in the domain - Be proficient with programming languages, VoIP, Asterisk interfaces (ARI, AMI, AGI), SIP configuration, API integration, and webhooks - Have robust problem-solving skills to provide innovative solutions Key responsibilities will include scrutinizing our present setup, identifying weaknesses, and implementing improvements to enhance call quality. A solid understanding of business requirements is vital to ensure the VoIP system is modified to suit our operational needs. Interested candidates, please email your resumes and cover letters.

I'm looking for an expert in VoIP/SIP setup, with experience in setting up contact centers, to help us with our inbound customer support setup in the US. The main requirements for this project are: - Setting up a VoIP/SIP system for inbound customer support for Indian callers - Managing this setup to accommodate more than 50 agents - Integrating the system with CRM software and a Ticketing system You should have: - Proven experience with similar projects - Expertise in VoIP/SIP setup and managing large contact center environments - Knowledge of CRM software and Ticketing system integration Your suggestions and recommendations are welcome.

...network simulation and coding to assist with my research project. My project is based on a computer network, WSN specifically, which needs to be accurately simulated and further customized. The network topology itself may be open to your suggestions, and I'm looking forward to discussing this in detail. Ideal Skills and Experience: - Expertise in Computer Network Simulation, specifically with NS2/NS3 - Strong coding skills to implement new variables into the simulation - Experience in research-oriented projects would be a plus. The successful completion of this project will involve developing a realistic network simulation, and then implementing custom variables to test specific research-related scenarios. The ability to create a detailed, efficient, and accurate simul...

...design for a barn workshop. This barn should have a large open studio area with lean-too on both sides. Mortis and tenon joints and dovetail mortis and tenon joints to be used every that a structural connection is required. Frame inside walls if studio space will be sheathed with 2x4 stud walls insulated with batt insulation with shiplap on lean top side and studio side. Roof will be capped with SIP panels. Roof will be metal and outside walls will all be metal siding. Key Requirements: - 3D Model and Design to be done in Revit or other approved software. - Per Michigan building code and ASCE 7 for design loading for SE Michigan. - All design files to be provided including 3D model, hand calcs, and pdf plan sheets with overall plan, elevation views, and individual member detai...

Configurar router y dispositivo para registrar ina linea SIP de O2

...framing details, mechanical, electrical, or plumbing plans, window or door schedules, a site plan, foundation plan, gutter or drainage plan, or 3D renderings. Since the exact plans you produce will not be inspected or built, such details are not important. The plans must only serve as a realistic and potentially buildable example of this kind of project, so I can get and compare estimates from SIP (Structural Insulated Panel) providers and builders about potential costs of constructing such a the building envelope. Therefore, while you are not specifying panels and construction methods, it would be helpful if you have an understanding of SIPs as used in US/Canada residential construction. You would only be hired after a video meeting to determine your ability to understand the...

We need to customize linphone's windows/macOS/iOS/android apps with regular PBX features, sms,sip Presence,BLF,etc. Needs to be customize with our company logo/colors. Change colors and logos to match our theme. Sleek and professional design. Login screen to accept username and password or scan QR code Incoming and outgoing calls. Call Transfer (Blind and assisted transfer) Hold Call Transfer Call Attended Transfer Call End Call Conference Call (merge the calls) Video calls Presence and IM features Call history Smart contact list, with address book synchronization for smartphones Audio/video conference calls and scheduled meetings Call transfer and multi-call management (pause and resume) One-to-one and multi-participant conversations Intuitive message de...