Asterisk pbx punët

Pershendetje Ardit, keni eksperience me Suite crm apo Vtiger crm? kam nevoje pernje konfigurim ne lidhje me crm dhe asterisk

I am looking for a developer to develop a call forwarding pannel and pbx panel on freeswitch voip. The call forwarding panel and pbx panel should include advanced call forwarding with scheduling and should include integrated voicemail and call recording. I am not sure which VoIP provider to use and would appreciate advice in this regard. I am targeting completion within a month. The successful developer will be knowledgeable and experienced in using freeswitch VoIP in order to complete the project. If this project is completed to my satisfaction, I will consider having the developer complete additional projects with me. Looking forward to hearing from an interested and qualified candidate. No asterisk Candidate. Only Freeswitch other voip developer only.

I have a VOIP PBX based on Freepbx with 2 extension . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is activated with the night and day option with code *280

I need asterisk to do the following: 1. asterisk Dial to callee 2. Callee pick up the call and the call is answered/ bridged 3. After X seconds, asterisk will inject DTMF tone 4. Caller will not be able to hear the DTMF.

Install 3CX PBX on digital ocean using dataphone as a SIP provider . You must have installed on digital ocean before or used dataphone as a SIP provider. You need to successfully get the 3CX working to complete the project. Regards

Hi Aqs Y., I'm looking for a consultant to upgrade from freeswitch 1.10.6 to the latest on debian 10 (most probably also debian needs to be upgraded, to 11 or 12, as you prefer) It needs to work with asterisk 16.29.0 and also needs to support mod_expr, ceil and randomize

We are seeking an experienced freelancer to facilitate the seamless integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and ...integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the utmost security for our call traffic through Encrypted Media. Project Requirements: -Integration of WebRTC functionality into the Isaaabel platform. -Configuration of JsSIP npm package to establish connections with our Asterisk calls to SIP and PJSIP extensions through the WebRTC interface. -Implementation of robust security measures, including encrypted media, to safeguard call communications.

I am seeking an expert in Plivo VOIP and PBX IVR configuration to assist me with my project. Specifically, I need assistance with setting up IVR menus and configuring call routing. For call routing, I do not have a preferred strategy, so I am open to suggestions. I need someone to setup to do IVR, setup greeting message, music on hold, voicemail, hours of operation routing and etc. If you are hired, I will provide Diagrams and prerecorded audio files. You must have all necessary skills and experience working with Plivo. Calls will be routed to a mobile app that you recommend unless Plivo has one. Ideal skills and experience for this job include: - Extensive knowledge of Plivo VOIP and PBX IVR configuration - Experience in setting up IVR menus - Proficiency in confi...

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I am looking for a freelancer who can help fix the issue with the Vicidial inbound carrier stats for my existing business. The specific problem we are experiencing is missing data in the reporting. We need this issue to be resolved within one day. Skills and experience required for this project: - Expertise in Vicidial and inbound carrier stats - Strong troubleshooting and problem-solving skills - Experience with data analysis and reporting - Familiarity with call center operations - Ability to work efficiently and meet tight deadlines

I am in need of assistance with a clean up and addition of features to an Asterisk PBX system. Specifically, I'd like assistance with setting up an IVR menu which can be used to direct customer calls to the right departments. Additionally, I would need the system to be maintained regularly after installation of the features. I have a preferred method of communication through email and am eager to find an experienced freelancer for the task.

I have been using a voip system that was setup by someone else. Recently it stopped working. I tried to login to fusion PBX but can't login anymore it says password not recognised. All other features of the system are working i.e domain name, telynx all ok need someone to help get it restarted

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I am currently facing an issue with my Issabel PBX setup connected to a D-Link DVG-6004S FXO Gateway. While I'm able to make outgoing calls successfully, I cannot receive incoming calls. Requirements: 1- Proficiency with Issabel PBX and D-Link FXO Gateways. 2- Ability to quickly diagnose and resolve the issue. 3- Availability to work on this immediately and resolve within 50 minutes. 4- Strong communication skills in English. If you believe you're the right fit, please contact me ASAP. Compensation will be provided for the quick and successful resolution of this issue.

Please don't bid if your bid request is more than 20 usd. We are looking for someone for installing Ubuntu Linux 22 and free pbx. Also configure apache for multiple domains and subdomain with latest php 8.1+. At the end provide a iso image that can be used to reinstall in any x64 machine.

I am looking for a skilled developer to create a Cloud-based PBX system with the following requirements: Features: - Call Recording capability - Auto Attendant for efficient call routing - Voicemail to Email functionality for convenient message retrieval Requirements: - The system should support more than 50 lines or extensions Ideal Skills and Experience: - Experience in developing Cloud-based PBX systems - Proficiency in integrating call recording functionality - Knowledge of setting up Auto Attendant and Voicemail to Email features - Ability to handle a large number of lines or extensions If you have the necessary expertise in building Cloud-based PBX systems with call recording capabilities and can handle a high volume of lines or extensions, please submit your...

I am looking for a freelancer who can help me configure my Grand Stream UCM 6304 IP PBX. I currently have both analogue lines and IP extensions in my setup. I require a specific configuration for my IP PBX and I am looking for someone who can provide seamless connectivity. Ideal skills and experience for this job include: - Proficiency in configuring IP PBX systems, specifically the Grand Stream UCM 6304 - Knowledge of both analogue lines and IP extensions - Experience in providing seamless connectivity for IP PBX systems If you have expertise in this area and can help me with this project, please submit your proposal. Thank you.

I am looking for a freelancer to help me integrate a WebRTC based calling experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encry...experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement so it would be great if the freelancer can deliver the project quickly. Resume, we need to connect WebRTC extension with Issabel. I need that JsSIP npm package can connect with my asterisk server and make calls to SIP and PJSIP extensions. I need the documentation of the implementation for future re-ins...

I am looking for an experienced Asterisk developer to create a web interface to manage all of its features. Specifically, I need call routing and forwarding, an Interactive Voice Response (IVR) system, and call recording and monitoring capabilities. No specific requirements for the web interface are necessary and the freelancer will not be expected to provide post-development maintenance or support.

...skills - Attention to detail for accurate data entry and call tracking Project Description: We are looking for a freelancer who can assist us with our Vicidial short duration project. The desired duration of each call is less than 1 minute, and the purpose of the calls is sales. We require someone with experience in using Vicidial or similar call center software. This project can be done on asterisk of vicidial web interface as well. Itshould be just a code on the dialplan. The specific features for the dialer have not been specified by the client. Therefore, we are open to suggestions and recommendations from the freelancer. The ideal candidate should have a strong background in sales, with proficiency in sales techniques and strategies. They should be able to ha...

I have a VOIP PBX based on Freepbx . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is engaged and in no other case.

I have a VOIP PBX based on Freepbx . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is engaged and in no other case.

I am looking for a freelancer who can help me deploy Asterisk 16 with a specific PJSIP configuration. The ideal candidate should have experience with Asterisk and PJSIP. Requirements: - Familiarity with Asterisk 16 - Ability to configure PJSIP according to specific requirements - Experience in handling concurrent calls, with a focus on optimizing for a single call The requirements are very simple. I have configured it myself before and achieved single-pass. If the extension calls the mobile phone, the sound of the mobile phone can be heard, but the sound of the extension cannot be heard by the mobile phone. You only need to configure the phone to be able to achieve dual communication! If the price is not suitable, the price can be negotiated as long as you can solve...

Project Description: Troubleshoot and resolve registration issues with PBX sip trunk - I am using an Asterisk PBX system and attempting to register a SIP trunk with a Telecom Provider - I am not sure if there are any error messages being displayed when attempting to register the SIP trunk - I am seeking a skilled professional who can help me troubleshoot and resolve any registration issues with the PBX sip trunk - The ideal freelancer for this project should have experience with Asterisk PBX systems and SIP trunk configuration - Knowledge of Telecom Providers and their registration processes would be beneficial

IVR PBX VICIDIAL INSTALLER I am looking for a skilled freelancer to provide a full installation of an IVR PBX VICIDIAL solution. Specific features that need to be supported include call recording, interactive voice response (IVR), and voice mail. The system will be used by a specific number of users, with the exact number to be determined. Ideal skills and experience for this project include: - Extensive knowledge and experience with IVR PBX solutions, specifically VICIDIAL - Familiarity with Asterisk and FreePBX - Ability to configure and set up call recording, IVR, and voice mail features - Previous experience working with systems supporting multiple users If you have the necessary skills and experience, I would love to discuss this project further.

I am looking for a skilled designer to create a logo for my new business. Company name is AJ Telecom we are into sales and services of all kind of telecom products like EPABX, IP PBX, Audio & Video systems please design and be able to bring my vision to life. The logo should be visually appealing and represent the nature of my business. Specific elements or symbols may be included, but I am open to suggestions in this regard as well.

I am looking for a freelancer who can assist with integrating 3cx PBX with Odoo. I currently have a 3cx phone system and an existing Odoo installation. The main task for this project is to integrate the two systems seamlessly. Skills and experience required: - Expertise in 3cx PBX and Odoo integration - Strong knowledge of VoIP systems and protocols - Experience with configuring and customizing 3cx and Odoo to meet specific business needs - Familiarity with CRM and ERP systems, particularly Odoo - Ability to troubleshoot and resolve any issues that may arise during the integration process The ideal freelancer will have a proven track record of successfully integrating 3cx PBX with Odoo and will be able to provide references or examples of previous work. They should a...

Necesitamos terminar un proyecto que se empezo, pero no se termino el desarrollo, utilizando para este proyecto vue 3 como front end y con backend asp.net core, utilizando la api de Twilio, la idea del proyecto es tener el control de llamadas que se hacen en la empresa, saber cuantas llamadas se hicieron y cuantas llamadas no se contestaron, llamadas devueltas, el proposito del sistema es tener un sistema administrativo para mejorar la atención de las llamadas de los clientes y tener un mejor seguimiento. La idea es hacer un sistema muy similar a este

...dont have Asterisk and SCAIP experience. I am in need of an English speaking Asterisk & Kamailio specialist to assist with my project. The specific tasks that I require assistance with include configuring the Asterisk and Kamailio servers and the settings. Specifically we want help in complying with SCAIP (Social Care Alarm Internet Protocol). If you have this experience that is perfect. We already have an Asterisk engineer but he may need some help with interpretation. I already have existing infrastructure in place, so the specialist will be working with that instead of setting up everything from scratch. The expected turnaround time for this project is moderate, ranging from 1 to 2 weeks. Ideal Skills and Experience: - Strong knowledge and experi...

I am looking for a freelancer who is experienced in FreePBX and can help me solve a problem with my caller ID. Currently, my caller ID is not displaying correctly. Skills and experience required: - Strong knowledge and experience with FreePBX - Expertise in troubleshooting and resolving caller ID issues - Familiarity with PBX systems and call routing - Ability to analyze and diagnose caller ID problems - Knowledge of SIP and VoIP protocols The ideal candidate will have a proven track record of successfully resolving caller ID issues in FreePBX systems. They should be able to quickly identify the cause of the problem and implement the necessary fixes to ensure that the caller ID is displayed correctly. Good communication skills and the ability to explain technical concepts to non-...

...though it does not necessarily have to be an animal; and we would like to place it in the triangle above the product name. We are looking for an organic font for the product name and something standard or type-faced for the tagline. The profile of the sacha inchi seed (the pulverized version being sold) would have preferred placement between "Ancient Seed" and "Modern Life" as if it were an asterisk separating the two phrases, or an abstracted version of it above the product name is an idea. In regard to color we have leaned toward the red with white boarder, and black, blue, white, and grey around images and fonts. We do, however, like the contrast of the logo against the triangle. That is a start, we are open to comment, question and collaboration. O...

We need our asterisk to play a small audio file to destination at the moment we receive 180 or 183 from destination. Destination provider needs 5 RTP packets to discover IP and port when asterisk is behind NAT.

I am looking for a skilled developer who can create a web front end for an e-commerce website selling Fusion PBX VoIP connections. The ideal candidate should have experience in building e-commerce websites and integrating secure payment gateways. The project involves creating a new website from scratch, so the developer should be proficient in web development and have a good understanding of user experience and design principles. Key functionalities for the e-commerce front end include user registration, allowing customers to create accounts and login to manage their purchases. Additionally, the front end should have product listings, showcasing the different Fusion PBX VoIP connections available for sale. The developer should also have experience in integrating secure ...

I am looking for a freelancer who can assist me with data entry work. Specifically, I need help with typing and data input. I already have the data that needs to be entered, so there is no need to worry about finding the data source. The volume of data is ...about finding the data source. The volume of data is relatively small, with less than 100 entries to be entered. Ideal skills and experience for this project include: - Proficiency in typing and data entry - Attention to detail to ensure accurate data input - Ability to work efficiently and meet deadlines - Familiarity with data management and organization The correct answers have an asterisk next to them just need to copy and paste across essentially I am up to question 17 follow the same structure and complete t...

...technology company specializing in advanced cloud PBX and SIP hardware solutions. We're seeking a skilled VoIP Expert to guide strategic decisions, offer technical expertise, and shape our cutting-edge communication offerings. Responsibilities: Advise on cloud PBX and SIP hardware solutions, ensuring seamless communication for businesses. Evaluate and recommend SIP hardware components, considering compatibility and performance. Create clear documentation for system configurations, troubleshooting, and maintenance. Lead hardware selection, vendor assessments, and cost-effective procurement. Collaborate with tech teams to integrate solutions into our existing infrastructure. Resolve complex technical issues related to cloud PBX and SIP h...

...language and dialect. Example: Portuguese (Brazil) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the Portuguese language, the file uses English words, then you need to put "Portuguese (Portugal), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your n...

...setting up Multi-tenant Fusion PBX gateway, Twilio Sip trunk, inbound/outbound routes, access controls, sip profiles and troubleshooting any issues that may arise as well as recommending some general best practices. Fusion PBX is already installed running on AWS and I have a basic knowledge of the system. Just looking for additional guidance over a video meeting. Requirements: - Familiarity with Fusion PBX and its configuration - Ability to set up the basic features and functions of Fusion PBX - Understanding of VoIP and telephony systems - Strong troubleshooting and problem-solving skills Ideal Skills and Experience: - Previous experience with Fusion PBX configuration - Knowledge of VoIP protocols and SIP trunking - Ability to customiz...

...we require integration of the apps with our existing ASTPP PBX server, with the following desired features: Support for audio calls with various codecs (g729, ulaw, alaw, Opus, iLBC, GSM) Video calls with video preview, supporting codecs such as H.264, H.265, and VP8 Management of multiple calls Call transfer, pause, and resume functionality Audio conferencing by merging calls into a conference Instant Messaging with end-to-end encryption for both one-to-one and group chat Capability to share pictures and files Address Book integration Call History display Presentation of advanced call statistics Echo cancellation feature Quality of Service optimization Ability to send and receive SMS text messages through the ASTPP PBX server Implementation of secure commu...

I have 4 people working from home including me, we do not have VPN, I do not have on-premise PBX system. However, I do have cloud PBX on Flynumber, Zadarma, and Bitrix24 for telephony, CRM, and everything else related to keeping the business running. A few months ago I purchased a GoIP4 SIP GSM Gateway, but been having a hard time configuring it, so it hasn't worked as expected up to now. I've got 4 local sim cards in the machine already but I never got them to work properly. I directed the calls to an SIP address on one of my virtual PBX, I want those sim cards to receive and make outgoing calls and I want all 4 people to be able to do so right from their stations at home even though we are not in the same network. At this point, I can receive calls but c...

I am looking for a freelancer who can integrate Issabel PBX with Vtiger CRM. The specific integration required is contact synchronization. Skills and Experience: - Strong knowledge and experience with Issabel PBX and Vtiger CRM - Expertise in integrating different systems and ensuring seamless data synchronization - Familiarity with telecommunication protocols and call routing - Experience in working with CRM systems and understanding of contact management - Ability to troubleshoot and resolve any issues that may arise during the integration process The integration needs to be done considering the existing system in place. Timeline: The preferred timeline for completing this integration is as soon as possible. If you have the required skills and experience, and can com...

Implement real-time phone speech recognition in project with Asterisk PBX and Kaldi/Vosk experience with Asterisk PBX and Kaldi/Vosk

...project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asteris...

...project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asteris...

i had 2 Yeastar S20 PBX in different countries the problem is between the 2 countries is blocked port's so the voice is not stable and you can't hear as well ,, we are using linkus for calling not VOIP phone ,, so is there solution as openvpn or something else ? or cloud VPN .. thanks

We need our asterisk to play a small audio file to destination at the moment we receive 180 or 183 from destination. Destination provider needs 5 RTP packets to discover IP and port when asterisk is behind NAT.

Se necesita tecnico experto para montar remotamente una central asterisk, que pueda hacer llamadas locales usando un modem 3g con chip celular. Configurar softphones para personal.

...======================================================= if you did a similar project before please share with us some pictures and information. you will be our priority. Hi, The WhatsApp channel allows consumers to connect with brands and brands to connect with consumers from the WhatsApp app on their smartphone. We need to develop a SIP to Whatsapp voice gateway to send some of the calls from our PBX. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with mul...

I am looking for a skilled freelancer who can help me change the operating system of my Openstage 20E Siemens IP Phone to SIP in order to work with Asterisk PBX. Key Requirements: - Experience in configuring IP phones and working with Asterisk PBX - Proficiency in uploading SIP firmware to the IP phone - Familiarity with Openstage 20E Siemens IP Phone and its operating system Specific Features: - Enable voicemail, call forwarding, and automated attendant on the SIP phone Preferred Method of Communication: - Remote access to the phone for configuration - Step-by-step instructions via email for guidance - Phone call or video conference for any necessary guidance If you have the necessary skills and experience to complete this project, please submit your propo...

We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/ all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.

Necesito instalar un conmutador virtual open source ISSABEL en un VPS para hacer llamadas entre extensiones y mensajes de texto, con salida de llamadas a la PSTN con troncales SIP.