Ivr asterisk mysql工作

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    13,222 ivr asterisk mysql 找到的工作,价格在 HKD

    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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    Asterisk has stopped running on my version of Freepbx. I do not know how to access the command line or the commands to restart. I just need someone to show me, preferably by doing a screen share and walking me through the steps on how to 1. access the command line on my server 2. enter the commands to stop, and restart asterisk

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    centralita 6 天 left

    Tengo un gateway goip gsm de 8 sim, 4 voipeadores tplink y un servidor en asterisk pero necesito crear la troncal sip y enlazarlos con los voipeadores para realizar llamadas salientes por el gateway gsm

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    I want to setup click2Dial function on my PHP website with Asterisk, the open source communications framework. There should be call button on press call should be the dial. and store the recording of that call. Have to setup Asterisk on AWS.

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    A2billing installation 6 天 left
    已验证

    I need to install a2billing in freepbx 13 with asterisk 11 and have to verified the command script by using teamviewer in my virtual local server. Kindly also need to show some configuration, like in freepbx IVR has two option and one of them is calling card option. so if customer chose the calling card option then how to divert them to a2billing and

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    Install & Setup Free PBX on Digital Ocean Droplet using Centos Setup with carrier, all basic features etc. I am new to this so will need walk thru after all is setup.

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    ...M., I want to install a new asterisk server. We will use it as a SIP proxy. We will have a blacklist B numbers in db (or anywhere), we will sync this data every minute.
And Asterisk will check if the incoming all is in blacklist, and will route the call after. The flow would be like: call comes from A server, to Asterisk server with 3333 prefix 3333XXXXXX

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    Need an ICE/STUN/T...have NAT WebRTC clients audio working fine with my Asterisk. Need to check and explain me how to configure Asterisk and WebRTC script (like doubango) to work when the client is behind NAT. I have coturn installed but not configurated. I see RTP packets but only in one way. Have server A with Asterisk and server B to WebRTC script.

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    Hi Toshazed, I need your help to install a2billing in freepbx 13 with asterisk 11 and need to show me the command script. it will be install in my virtual local server. Kindly please help me to show other configuration which will be discuss after installation.

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    Conexion CRM Bitrix e Issabel 5 天 left
    已验证

    ...conectar nuestra Central telefónica Issabel (Asterisk) con el CRM Bitrix 24. Lo que queremos lograr es poder realizar llamadas telefónicas desde el mismo Bitrix a través de las centrales Issabel ya configuradas. El bitrix cuenta con instructivo para conectarlo con FreePBX. Si bien este se maneja también con Asterisk, es necesario hacer la configuraci&oa...

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    I need some help to fix up some call problem with an Asterisk server - All internal are behind NAT (with Miktotik router) - Asterisk server are on a VPS on cloud

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    ...response (IVR) dengan kecerdasan bantuan untuk generasi seterusnya dalam sektor servis khidmat pelanggan. Matlamatnya adalah untuk mesin berinteraksi dan bertindakbalas kepada pertanyaan pelanggan dalam segala senario harian, seperti perbankan, hospitaliti, dan lain-lain lagi, tanpa input ataupun input yang minima. Anda boleh baca mengenai sistem IVR dengan

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    We are looking for a core PHP expert with experience in OOPs, MVC, 3 tire architecture, video & audio chat using asterisk and e-commerce for online training purpose.

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    ...We have a current very old custom build solution based on a Asterisk PBX with ISDN and want to move it to the cloud and provide new features for our team. Its very important that you have indept knowledge arround Asterisk PBX and integration with SQL / External scripts from within Asterisk PBX We can accept that you just provide the SQL and we can

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    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    We require the following 1. API for existing dialer 2. New Dialer with CRM and IVR Interested concern please submit your bids for further discussion.

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    Need someone to remotely help me with configuring an IPPbx solution. Consists of: 1 HA100 2 UCM2510 100 extensions 4 conference rooms Single-level IVR configuration 2 SIP trunks configuration Write a manual to start a conference room Write a manual to add an extension Write a manual to make a backup I need someone to be able to provide remote support

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    IVR message for a new call center

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    We need a server specialist that can install tonight (with information from our developer) an Asterisk VoIP Server on a CentOS. Next to installation and proper working of software we need you also to increase the security settings of the server. Budget $100. Must be done in one day.

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    Looking for someone who are expert in fixing the asterisk server and goautodial server to fix some issues. I am trying to connect that server to external public IP behind the firewall/router. Somehow there are some errors such as " handle_request_register: Registration from '"1300" <sip:1300@[登录来查看链接]>' failed for '[登录来查看链接]' - Wrong password

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    I have a Asteris 11.14.1 PBX. I want to use Bitrix24 CRM and log all incoming and outgoing calls in the CRM. I need to connerct Asterisk with Bitrix24 for inbound and uotbound calls

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    I am setting webrtc to an Asterisk server,I need an engineer with experience in networks and Asterisk webrtc configuration, in order to fix error net::ERR_CERT_COMMON_NAME_INVALID. I have a Goddady certification. Asterisk show TLS/SSL certificate ok.

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    ...} If I request this web address, Asterisk will: 1-) Call {CALL} via {CALLERID}. 2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service. - Need support over 50 calls at same time. - I have a server at digital ocean. Asterisk is not installed. - I have a SIP account

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    Hi, we are looking someone (Female only) who can record an prompt (IVR GREETING MESSAGE) in English for our company thanks

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    ...solve the pages does not update correctly the Vicidial/Asterisk MySQL tables. I need a correctly full working installation, the result should be an installable and fully working ISO or Virtual Machine harddisk with the working installation. Requires: dkms-dahdi, asterisk, vicidial, php7, MySQL, MariaDB, CentOS, and other of the same kind. I need

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    asterNET modify sip trunk 1 天 left
    已验证

    asterNET modify sip trunk need a vb.net or c# to modify sip trunk in asterisk from a windows base software

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    We need to integrate an asterisk pbx with SuiteCRM if possible with freepbx or raspbx Integration means: - On inbound call open CRM calling contact information - Clicktocall from CRM - Logg inbound and outbound calls in CRM

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [登录来查看链接] Eventually another

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [登录来查看链接] Eventually another

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    Hello All, I'm looking an experienced Microsoft system admin. The freel...windows server 2016 needs prior to be prepared before the installation of Exchange server 2016. UC must be configured as well since we will also be hosting VoiceMail for an Asterisk VoIP system Note: Apply on this job only if you already perform such as job. Regards, Léon

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    Appen is kicking off a large scale speech collection project to help our clients to build the next generation of artificial intelligence powered interactive voice response (IVR) systems in the customer service areas. The end goal is to allow machines to interact with and respond to customers with their inquiries in any daily scenarios, such as banking

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    We need someone that could write an interface to make sure that Asterisk and a Hotel Management software, Opera (that supports FIAS Protocol) can communicate, especially the billing of the calls, wakeup call, caller id.

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    Project for Sourish B. 已经结束 left

    ...following project: Hi, I read your gig and would like to share with you what kind of project we have in my company in Bulgaria. I am a little bit confused about PBX, VoIP, Asterisk therefor I will describe what we need as functionality. Right now in the company there are 3 hard phones, one landline number 00359 (0) 52 630 540 and one fax number 00359

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    Create a Survey FreePBX/Asterisk module to measure customer satisfaction levels for our call center agents services. Current software versions: Elastix 4 Here is what is needed. After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 1 or more fixed questions. We

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    I have a personal asterisk server for my self and kids and I see lots of messages like this. --STATUS: HANGUP on h from from ,,XXXXXX") in new stack Looks like some one trying to make calls but I dont have outgoing plans just a few extensions. I have fail2ban installed but it doesnt seem to be catching them.

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    ...fields ... the only field I have doubts about is the one labeled as follows: The difference between your server time and the [登录来查看链接] server time (in seconds)*: The asterisk (*) at the end of the field name corresponds to the following: * Please ensure your system clock is set to proper time and time zone. AuthorizeNet gateway will validate the

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    Hi! I need help with Freepbx CRM and asterisk API's: I need customers to be able to make a call from the CRM on the browser with a click on a button using an API, not with Zulu software or plug in.

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    Asterisk Dialplan 已经结束 left

    I have asterisk server with whitelist number and make any number hit my asterisk to pass on this whitelist and if number in whiltelist it pass if not it hangup . now i need to add some options for non whitelist numbers like playback press 1 to continue and if he press add it to my whiltelist also need to send 503 code when call hangup after not passing

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    When calling the business main number I will like to have Background music, Talking about our service and promotion. Needs to be a loop average 1 -3 minutes.

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    ...intelligence powered interactive voice response (IVR) systems in the customer service areas. The end goal is to allow machines to interact with and respond to customers with their inquiries in any daily scenarios, such as banking, hospitality, and more, with no or little human input. You can read more about IVR systems here. In order to do this, we are collecting

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    we need Asterisk Call Center System developer to work with us project base .. with full experience of : - Basic Call System - Call Center System - Call Rating System - Call Details Reports - Branding The system for us .. - CRM Integration

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    Asterisk/FreePBX Genius! 已经结束 left

    I need the full Amazon AMI Install with Freepbx tweaking in order to make this 100% functional. [登录来查看链接]

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    Trophy icon WebRTC help needed 已经结束 left

    I use [登录来查看链接] v0.11.6 for my WebRTC project. Everything works well but there is no sound on reciever side. Only caller can hear. Asterisk 16 is installed on Centos 7. Need help to fix the issue.

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    Seeking a professional female voice actor to read our IVR script. American native speakers only. Read should be well enunciated and enthusiastic. I will attach an example read for cadence reference.

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    Setting up an IVR for me and call and email blasting from countries like US,UK,CANADA etc

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    ...den Chat fragen. For English project: Task Overview Appen is helping clients to build the next generation of artificial intelligence powered interactive voice response (IVR) systems in the customer service areas. The end goal is to allow machines to interact with and respond to customers with their inquiries in any daily scenarios, such as banking

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    We have 3 issues with this integration: 1- For the outgoing calls, the popup shows the trunk ID number instead of the dialed number. 2- When rece...When press on the call icon in the web browser, the popup show up but the extension is not calling. Project is on high priority, Please only bid if you have experience in Asterisk with Zoho Integration.

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    Speech Collection Project 已经结束 left

    Task Overview Appen is helping our clients to build the next generation of artificial intelligence powered interactive voice response (IVR) systems in the customer service areas. The end goal is to allow machines to interact with and respond to customers with their inquiries in any daily scenarios, such as banking, hospitality, and more, with no or

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    ...see: The numbers of fields that has to sign on the bottom (fixed) Only the fields that has to sign with a yellow background Red borders on the bottom and an asterisk if required The text will fit inside the cell if longer will be resized. If the user use the desktop, can sign using the mouse. When the user click the save button, the

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