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    15,281 asterisk alsa buffer tugasan ditemui, harga dalam USD

    Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

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    Hi We are looking for a freelancer experienced in Asterisk. Current developer works at another job, so you will work with me for a long term if you want. hourly rate is 25~35. 40 hours per week Thanks Anthony

    $24 / hr (Avg Bid)
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    This is an on-going project with various tasks managing asterisk Please apply only if you have experience in this.

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    1- I want to have the same navigation that is on current home page but on [log masuk untuk melihat URL] , [log masuk untuk melihat URL] [log masuk untuk melihat URL] , [log masuk untuk melihat URL] , [log masuk untuk melihat URL] so all the sites are homogeneous , and they all belong to healthsonar.com. Some sites do not have navigation so please add it to the Top navigation. Does not have to be...

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    hello this is for [log masuk untuk melihat URL] 1- I want to have the same navigation that is on current home page but on [log masuk untuk melihat URL] , [log masuk untuk melihat URL] [log masuk untuk melihat URL] , [log masuk untuk melihat URL] , [log masuk untuk melihat URL] so all the sites are homogeneous , and they all belong to [log masuk untuk melihat URL] 2- speed up the site ...

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    FreePBX predictive dialer 6 hari left
    DISAHKAN

    I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not ...

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    Actualmente tenemos ya funcionando una centralita Asterisk con FreePBX y queremos implementar un sistema de encuestas telefónicas para valorar la atención telefónica que damos a nuestros clientes. El funcionamiento deseado podría ser UNA de estas 2 opciones: - Opción 1: tras hablar con uno de nuestros agentes, la llamada pasa automáticamente a una I...

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    Hello, We looking to make form builder, either drag and drop or just have field option and once admin will select then it be added in the form. - need to have option to add custom background color - need to have option to change button color option as custom colors. - once form submitted then form need to save in the DB and you need to show in the admin area form wise report. - also once sub...

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    We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a pub...

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    Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database

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    Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall...

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    7 bida

    Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk . a strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk. Hint: the Freepbx is working on AWS EC2 instance.

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    Linux-Huang 5 hari left

    Design a programming solution to the bounded-buffer problem using the producer and consumer processes

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    Buffer Overflow Download project-part1 and, using a buffer overflow, determine the expected input that will produce the "Correct!" message.

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    Buffer Overflow 4 hari left

    Download final-part1Preview the document and, using a buffer overflow, determine the expected input that will produce the "Correct!" message. Submit the expected input and describe in detail what constraints cause the "Correct!" message to be displayed, as well as your process for solving this. NOTE: You must disable address randomization in order for you to complete this ques...

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    1: Buffer Overflow Project-part1, using a buffer overflow, determine the expected input that will produce the "Correct!" message. 2: Shellcode Produce an assembly shellcode listing that will read 10 characters from standard input and write the same 10 characters to standard output. You need only to produce the assembly listing, and not the assembled bytes or shellcode sample program. ...

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    Looking for Asterisk, FreePBX with WebRTC specialist to verify and fix existing PBX. This is no place for amateurs !

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    Restore Asterisk PBX 4 hari left
    DISAHKAN

    I have a running asterisk PBX - i will to rebuild new one with asterisk. i use Asterisk API, Databases.

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    Developing an 8 channel EMG system using an ADS1298, the hardware for the project has been designed, now the firmware needs to be developed. The design consists of 8 monopolar AFE channel connected to the ADS1298, the data is collected by a microcontroller (MCU) and send to the PC to be visualized (MATLAB GUI). I only need the firmware for the MCU to be developed, as I have tried multiple time but...

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    Asterisk with python agi 3 hari left
    DISAHKAN

    Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action

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    migration of asterisk to clean server

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    migration and debugging asterisk

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    migration to asterisk 16 some scripts to get up and running

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    I am migrating my asterisk server to latest . I need some help to resolve the issues

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    we need an expert in Networking to create a VPN between 2 servers 1. both servers are a VM in a windows 2. Server #1 we have asterisk that need to allocate an Public ip [log masuk untuk melihat URL] #2 is connect to local GSM Gateway to test it we need to make a call via the asterisk using the vpn and ending up at the gs...

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    Perfil profesional: Informático, Programador, Ingeniero Telecomunicaciones, Matemático Conocimientos técnicos (Experiencia mínima 5 años en cada una): - JAVA - PHP - JAVASCRIPT - LINUX - Gestion de sistemas y redes - Bases de Datos (MySQL y SQL) - Experiencia en servidores Otros conocimientos técnicos: - ERPS y CRM - Android o IOS - Asterisk y/o FreeP...

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    Add On My EA 2 hari left
    DISAHKAN

    (1) I need an input option called "Bulls" This input option will send orders on BULLISH candle's HIGHS when a bullish engulf or idx candle appears on the chart while the original strategy of the EA continues to also send orders on the bullish candle's low. (2) I also need an input option called "Bears" This input option will send orders on BEARISH candle's L...

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    Resolve issue with “Cannot Conner to Asterisk” error. Update server and enhance security.

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    I need someone to help configure the Patton SmartNode 4114 (FXO Port) with Freepbx/Asterisk so we can use SmartNode 4114 as a VoIP gateway for PSTN lines.

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    Online Ad promotion -- 2 2 hari left
    DISAHKAN

    Process Details: As ads are primary objective in this marketing field for every business. Well it’s a easy campaign where a processed center will be getting a individual portal and they have to access to their own login at any time and they need to watch the ad video for at least 15 seconds. These process must be done confidently as the agent would be required to login and see the video on ...

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    we need to get support for any one who know the chan_dongle and asterisk good

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    WebRTC to SIP gateway 2 hari left
    DISAHKAN

    I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls usi...

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    Online Ad promotion 2 hari left
    DISAHKAN

    Process Details: As ads are primary objective in this marketing field for every business. Well it’s a easy campaign where a processed center will be getting a individual portal and they have to access to their own login at any time and they need to watch the ad video for at least 15 seconds. These process must be done confidently as the agent would be required to login and see the video on ...

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    We’ve got an asterisk system with two trunks. We need some extension configuration changes and some ongoing support

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    Hi there! Looking for a SEASONED marketing assistant/ project manager for a marketing agency that can do the following: - Research marketing technologies like social media tools, email software, landing page software, etc and come back with the best ones based on a list of prerequisites. - Complete templates to send to other freelancers like bloggers and developers explaining tasks that need to ...

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    Ditampilkan
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    I want to change A2billing AGI to FastAGI due to performance and scalability. I need a very experienced person with a2billing and ofcourse asterisk.

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    I wrote a Script that returns a True or False boolean according to a lookup number from a website. I need someone to write an AGI on my [log masuk untuk melihat URL] on my Asterisk server, whose main purpose is to forward a Dialed Number to my Script in Python, and afterwards if the number dialed is returned with True, then the call should be allowed to go through. If the number is returned with F...

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    hello, we use a "less secure app" with our Asterisk PBX voicemail to email message notifications using gmail. gmail wants users who use this option to make them more secure. if you google "less secure apps" and gmail / gsuite you will see what needs to be done. this is what needs to be done: The G Suite Team <gsuite-noreply@[log masuk untuk melihat URL]> Tue, Jul 30, 12:...

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    9 bida

    Hi Mia, I noticed your profile and would like to offer you my project. We are Recruitment Agency and planning to launch a website (3 pages: Home, About Us & Contact Us) and print out brochure (2 pages content). We are searching for copywriter to help us write the contents. Both contents will be very similar whereby the website will have complete information and brochure contents will be the si...

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    1 bida

    Hi, I noticed your profile and would like to offer you my project. We are Recruitment Agency and planning to launch a website (3 pages: Home, About Us & Contact Us) and print out brochure (2 pages content). We are searching for copywriter to help us write the contents. Both contents will be very similar whereby the website will have complete information and brochure contents will be the simpli...

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    1 bida

    Convert files from wav to mp3 files after a call is made, historic data and new data automatically after a call is made. when change is made I want to see and hear from crd reports. review and clean log data from /var/log/asterisk make rule to minimize log file size /var/log/asterisk/fail2ban /var/lib/fail2ban /var/lib/fail2ban/[log masuk untuk melihat URL]

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    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

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    call Termination Expert 16 jam left
    DISAHKAN

    we need an expert in call termination that have expertise in 1. DINSTAR equipment [log masuk untuk melihat URL] VPN 3. and avoiding DPI 4 asterisk we need to set up a rout

    $246 (Avg Bid)
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    I started to build this web based SIP phone using [log masuk untuk melihat URL] - [log masuk untuk melihat URL] - but other work has left the project incomplete. I need the project completed and updated to use the latest version of [log masuk untuk melihat URL], 0.15.6. You will be provided with FTP access to the current source files including the HTML, CSS and current JS files. Additionally 3 S...

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    Hi Melody, I noticed your profile and would like to offer you my project. We are Recruitment Agency and planning to launch a website (3 pages: Home & About Us) and print out brochure (2 pages content). We are searching for copywriter to help us write the contents. Both contents will be very similar whereby the website will have complete information and brochure contents will be the simplified ...

    $181 (Avg Bid)
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    1 bida

    Hi Jocelyn, I noticed your profile and would like to offer you my project. We are Recruitment Agency and planning to launch a website (3 pages: Home & About Us) and print out brochure (2 pages content). We are searching for copywriter to help us write the contents. Both contents will be very similar whereby the website will have complete information and brochure contents will be the simplified...

    $181 (Avg Bid)
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    1 bida

    Hi Clarence L., I noticed your profile and would like to offer you my project. We are Recruitment Agency and planning to launch a website (3 pages: Home & About Us) and print out brochure (2 pages content). We are searching for copywriter to help us write the contents. Both contents will be very similar whereby the website will have complete information and brochure contents will be the simpli...

    $181 (Avg Bid)
    $181 Avg Bida
    1 bida

    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

    $378 (Avg Bid)
    $378 Avg Bida
    8 bida

    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

    $466 (Avg Bid)
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    3 bida

    I am looking for someone who has a good command of the English Language. I have 100 Picture Captions which I need jacking writing for, on the platform Buffer. The writing needs to attract the audience by making them think and finally asking questions so as to make the reader engage by posting comments underneath. See the example attached. I am more interested in someone who can write engaging ...

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