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    11,440 asterisk alsa plughw 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.

    $1553 (Avg Bid)
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    3 bida

    We need installation of newfies dialer, to get 2000+ concurrent calls from each box, it should be build on VM enviorment. Also asterisk amd module and beep detection modules need to be install.

    $50 / hr (Avg Bid)
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    3 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 or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.

    $88 (Avg Bid)
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    14 bida

    Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will not be considered.

    $2446 (Avg Bid)
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    Java application 5 hari left

    Create and test an application in java that will allow a user to create a text file to store data values associated with Player objects. The application will provide a menu for a user to display an encrypted list, display a decrypted list, add, update, and delete the Player data record in the text file. The menu will also contain a help command for the user to display the menu and an exit command ...

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    Trophy icon ASTERISK AMI 5 hari left

    Hello, we need asterisk AMI script (syntax) for yeastar PBX we need functions must work via triggering AMI commands described and tested : 1. hangup 2. mute 3. attended transfer 4. hold please only serious freelancers with experience. Please be aware that Yeastar PBX has limited manager commands

    $260 (Avg Bid)
    Dijamin
    $260
    1 penyertaan

    Hi, I'm having lots of problems with FreePBX. Trying to update: Unsupported Version of Asterisk, You need at least 11.11.0 you have 11.8.1 Running Amportal: amportal a ma refreshsignatures Fetching FreePBX settings with gen_amp_conf.php.. /usr/local/sbin/amportal: line 52: export: `®': not a valid identifier /var/lib/asterisk/bin/freepbx_engine: line 100: export: `&re...

    $22 / hr (Avg Bid)
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    9 bida
    Help for Asterisk-FreePBX 4 hari left
    DISAHKAN

    I have an Asterisk-FreePBX System with multiple disks that needs some fixes. If You are a specialist in this field, lets talk. --- This is not for people who think all the answers are on the Internet ! This is for experienced specialists. Requirements: Asterisk, FreePBX, SSL Certificates (Letsencrypte), Apache, multiple disks on system, web dev, PHP, etc. Will divide into milestones.

    $366 (Avg Bid)
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    Dear Coders, We are looking for a programmer who is familiar with Voip Systems and with the following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real Time C...

    $2848 (Avg Bid)
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    Hello, I have an asterisk PBX vers 11.22.0 . I am using a Polycom sound point IP 650. All works fine except for the transfer button. The transfer on polycom use SIP REFER to transfer the call. This is not working. Need help from anyone who know about the subject. Please respond to this project with "What up Dingo" at the beginning of your message so that I know you have read.

    $145 (Avg Bid)
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    3 bida
    speech to text 2 hari left

    Hello I have Asterisk dialer and I need to set up speech to text transcription (ONLY) I use to use to use IBM watson api for this, but it has become too pricey. it is 1 Min length audio of ivr recordings each. But total millions of files. every 2-3 months 7 million 1MB, 1 Min audio files.

    $52 / hr (Avg Bid)
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    15 bida

    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

    $22 / hr (Avg Bid)
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    28 bida

    This is an on-going project with various tasks managing asterisk Please apply only if you have experience in this.

    $32 / hr (Avg Bid)
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    15 bida
    FreePBX predictive dialer 2 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...

    $563 (Avg Bid)
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    10 bida

    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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    7 bida

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

    $592 (Avg Bid)
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    14 bida

    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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    9 bida

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

    $347 (Avg Bid)
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    9 bida
    Restore Asterisk PBX Akan Tamat left
    DISAHKAN

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

    $278 (Avg Bid)
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    10 bida

    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

    $508 (Avg Bid)
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    12 bida

    migration of asterisk to clean server

    $24 / hr (Avg Bid)
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    1 bida

    migration and debugging asterisk

    $34 / hr (Avg Bid)
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    1 bida

    migration to asterisk 16 some scripts to get up and running

    $28 / hr (Avg Bid)
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    1 bida

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

    $25 / hr (Avg Bid)
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    1 bida

    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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    2 bida

    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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    40 bida

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

    $150 (Avg Bid)
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    1 bida

    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.

    $175 (Avg Bid)
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    5 bida

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

    $68 (Avg Bid)
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    3 bida

    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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    10 bida

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

    $13 / hr (Avg Bid)
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    9 bida

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

    $388 (Avg Bid)
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    5 bida

    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...

    $156 (Avg Bid)
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    10 bida

    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

    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]

    $144 (Avg Bid)
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    12 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...

    $394 (Avg Bid)
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    6 bida

    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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    5 bida

    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...

    $754 (Avg Bid)
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    28 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

    Instalar asterisk y configurarlo para conectar con odoo

    $15 / hr (Avg Bid)
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    11 bida

    Hi Ibrahim Ali M A., I noticed your an expert in VOIP and asterisk. We are having issues with our VOIP system - in particular outgoing calls through SIP trunk are getting cut off in 6 minute 39 seconds. Asrerisk server running on CentOS. Can give access through SSH.

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    1 bida

    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

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    20 bida

    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

    $153 (Avg Bid)
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    2 bida

    I’m looking to setup a Hosted PBX company and would like help from a developer who has experience performing this service.

    $577 (Avg Bid)
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    20 bida

    Hace un mes me crearon un panel que puedo acceder por web y junto con Issabel Asterisk me registran las llamadas que se hace y lo muestra por la web. Hace unas semanas todo funcionaba perfectamente al 100%, al cabo de 2 días me dio el mismo fallo que muestro aquí, y el creador me lo arreglo, pero al cabo de unos 2-4 días dejo de funcionar de nuevo, y ahora me dan largas y no l...

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    6 bida

    To consolidate our different projects, we decided to write our own backend service to Asterisk PBX by utilizing the AGI specification (see https://wiki.asterisk.org/wiki/display/AST/AGI+Commands ). Because of performance reasons, this back-end should be implemented by using plain C, with as less external libraries as possible. We aim to use this service on a broad range of hardware, so it is imper...

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    6 bida

    We need to setup the integration with Asterisk 13 and Google Dialogflow to connect the Dialogflow bot to our telephony service. We have test environment with Ubuntu 16.** server with installed Asterisk 13. There are some add-on to make that integration: 1. [log masuk untuk melihat URL] 2. [log masuk untuk melihat URL]

    $130 (Avg Bid)
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    2 bida

    We need to setup the integration with Asterisk 13 and Google Dialogflow to connect the Dialogflow bot to our telephony service. We have test environment with Ubuntu 16.** server with installed Asterisk 13. There are some add-on to make that integration: 1. [log masuk untuk melihat URL] 2. [log masuk untuk melihat URL]

    $683 (Avg Bid)
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    3 bida