Sip nat traversalproiecte

Salut, Pe lan 1 avem reteaua locala calculatoare si telefoane voip si internet Pe lan 2 avem acces la serviciul digi sip care la care sa conectat dar nu mai vedem telefoanele din retea initial vedeam telefoanele ba chiar am gacut si teste cu un modul GOIP care a functionat bine .

Am nevoie de cineva sa imi configureze un telefon cisco 7960 pentru serviciul de sip.

Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

Salut, Am un VPS cu FreePBX functional. Doresc sa adaug HylaFax + AvantFax si sa pot sa trimit faxuri prin iax sau sip. Am vazut ca ai facut ceva similar. 40$ fixed rate Poti sa ma ajuti?

Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

Sunt interesat sa discutam pentru un proiect android, de voip (sip) asa cum este si viber sau skype... in primul rand ma intereseaza aplicatia pentru client dar si partea de server ca aplicatia sa fie functionala

Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

Ma intereseaza sa discutam mai multe legate de o aplicatie de voip pe android ca si viber , caut pe cineva care sa faca partea de client sau sa customizeze un client open source de sip existent

Buna seara, as vrea sa discutam cu privire la o aplicatie de voip in care utilizatorul sa se logheze pe baza de nr de tel si sa primeasca sms pt confirmare precum si prin facebook, sa aibe o parte din functiile viber, configurarea SIP sa se faca printr-un api fara ca sa trebuiasca sa faca ceva utilizatorul final pentru a configura la inceput aplicatia.

...functie de cum iti este mai usor) Fereastra in care se introduce parola are, sub campul de text, buton OK si Cancel, ambele fara functii atribuite. Adevaratul OK va fi localizat, invizibil, altundeva. Telefonul se inregistreaza prin 3 parametrii: Server Username Password (campurile username si password devin hash-uri si nu pot fi vazute in clar) Telefonul functioneaza atat pe protocolul SIP cat si pe IAX2. Telefonul poate deschide oricate linii in paralel. Eventual, fiecare linie noua deschisa, le trece pe celelalte pe Hold, in sensul ca se aude numai linia activa. Atunci cand se inchide o linie, din oricare motiv in afara de apasarea efectiva pe butonul Hang-up, telefonul incearca sa o redeschida la infinit (sau de un nr de ori care poate fi setat din men...

...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 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.

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 ..

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.

...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...

I'm seeking a seasoned professional to provide ongoing support for my existing Fortinet Firewall. The ideal candidate will have a strong grasp on the following areas: - **Configuration Adjustments**: I require assistance in fine-tuning the firewall rules, NAT settings, and VPN configurations. It's crucial that the candidate is well-versed in these areas, as even minor misconfigurations can have severe security implications. - **Troubleshooting Connectivity Issues**: The freelancer should be proficient in identifying and resolving connectivity issues. Quick and efficient troubleshooting skills are paramount to keep my network running smoothly. - **Implementing New Security Policies**: I need to ensure that my security policies are always up-to-date. The candidate should ...

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...

My SIP Trunking system is constantly preventing me from receiving inbound calls. It's a constant issue that requires urgent attention. Ideal candidates will possess: - Extensive experience with SIP Trunking - Knowledge in diagnosing and resolving call reception problems - Proficiency in troubleshooting technical issues constantly, not sporadically.

... This configuration should allow me to deploy the EC2 instances and also bring-down when needed. Key Requirements: An EC2 instance with Security group, Keypair, VPC components, CloudWatch logs and WAF integration. The EC2 will be launched in Private Subnet, with ALB and no direct traffic allowed to the instance. A Bastion/jump server for SSH to EC2. EC2 connects to the internet using NAT-gateway. WAF to be integrated with ALB, and ALB only listens on HTTPS. Security groups are restricted with limited access. Allocate and associate an elastic IP to the instance. The EC2 instance will run Ubuntu and serve as application server. I can provide AMI image of the web-app. It is a Python FastAPI/Nginx app. No other service is needed like RDS or S3 etc. Docker or Kubernetes are not need...

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.

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...

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...

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.