Viber sip asteriskproiecte

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 .

Salut! Suntem producatori din Ucraina, cautam parteneri de afaceri in Moldova si Romania. Scrie-ne pe viber sau whatsapp, discutam toate detaliile +380685599807. Toate detaliile pe site-ul

Salut! Ma poti ajuta cu instalarea unei platforme mailwizz? Ma poti contacta la whatsapp sai viber la numarul 0746797977. Multumesc!

Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!

Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,

Asterisk script

Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.

Bună, Razvan A.. Ți-am remarcat profilul și vreau să-ți ofer proiectul meu. Putem discuta detaliile într-o conversație privată. Putem discuta detaliile într-o conversație privată. +37379976677 / whatsapp, viber, telegram. Este vorba despre un proiect de copywriting - 80 de articole in limba romana. Te rog sa revii, daca esti cointeresata.

Bună, Mihaela M.. Ți-am remarcat profilul și vreau să-ți ofer proiectul meu. Putem discuta detaliile într-o conversație privată. +37379976677 / whatsapp, viber, telegram. Este vorba despre un proiect de copywriting - 80 de articole in limba romana. Te rog sa revii, daca esti cointeresata.

Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.

Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de 1000$.

Buna ziua, m-am uitat peste profilul dvs. si am observat ca aveti cunostinte in "VOIP (Asterisk)". Doresc o aplicatie in care un numar de telefon de exemplu "+40728182013" sa primeasca apel dupa apel la un interval de maxim 10-15 secunde, dupa fiecare apel primit sa se schimbe numarul de telefon care apeleaza pentru a nu fi blocat, iar daca are casuta vocala numarul "+40728182013" sa nu mai apeleze. Am un buget de maxim 1000$

Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk de la RDS

Salut, am instalat goautodial care include si asterisk, am reusit sa trec peste mai multe probleme succesive pana m-am impotmolit din nou. Cand incerc sa initiez apeluri imi spune ca nu este prefixul corect. Eu cred ca este din Dialplan Entry:, m-am uitat si la logurile CLI dar nu gasesc nici o solutie. Ma poti ajuta? Am un abonament de siptrunk la RDS

Configurare campanii, db, si securizare asterisk, goaudiodial, vtiger.

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

Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

Salut poti lasa un icq jabber yahoo daca ai te rog. Am nevoie de ceva pentru asterisk................

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

Vreau o apliactie de voip cum este si viber, crezi ca ai experienta necesara pentru a face una? Da-mi un replay pentru a discuta mai multe.

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.

Vreau o apliactie de voip cum este si viber, crezi ca ai experienta necesara pentru a face una? Da-mi un replay pentru a discuta mai multe.

...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...

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...

...highly-skilled Asterisk VoIP specialist for a unique project. I require expertise in Asterisk integration, VoIP configuration, and particularly in IVR implementation. KEY REQUIREMENTS: - Asterisk Integration: Full integration set up and testing needs to be completed. - VoIP Configuration: This includes configuring phone lines, extensions, and all essential VoIP functions. - IVR Implementation: The main purpose of this project is the implementation of automated IVR menus. This needs to be user-friendly and functional. The system should be capable of handling less than 50 concurrent calls. IDEAL CANDIDATE: The successful freelancer must have outstanding experience in all mentioned areas with a focus on IVR implementation. Any proven record in developing low-volum...

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.

We're looking for a talented person from The Philippines to join our Shopee team as a live-seller on Shopee Philippines. As a live-seller, your main job will be to connect with customers and show off our products in a fun and convincing way to boost sales. This is a work-from-home position where you'll...to do live-selling sessions every day for 6 hours (sessions can be spread out throughout the day). You will receive support and training if and when necessary to help supplement your growth and success. You'll get a monthly salary of $300, plus chances to earn more through commissions based on your sales and performance. If you are interested in this position, we would be happy to call you via Viber, WhatsApp, Messenger or any other platforms to help facilitate the h...

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

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...

...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 SI...

...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. ...

European Portuguese (Portugal) natural conversation recording. It is free talk conversation ( no script) i will give you only Topics and you both speak according to topics. please mention these detils in your bids *Are you native European Portugues from Portugal? *did you attache one minuet group audio sample on y...application and making the phone calls simultaneously. The recorders shall arrange two phones in total One phone is downloading Our Application The other phone is to dial the phone number simultaneously. We'll give you IDs and the phone number once you're ready. Note-The phone call may incur some telephone bills. So it's better to use any apps that can dial our Chinese phone number like Skype or viber cell package or it's cheaper if you have v...

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...

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 SI...

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...

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.