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    21,751 asterisk voip server configuration redhat linux 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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    I have Configured SSL in my Tomcat, my application is Spring MVC. The SSL set up is not working on Server.

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    ...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....

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    ...with the delivery of mail: Some arrive at the SPAM tray, others do not arrive and others simply bounce. And these bring us many complaints from our customers. Hello, In our server we have problems with the delivery of mail: Some arrive in the SPAM tray, others do not arrive and others simply bounce. I often receive calls telling me that: - The emails

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    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

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    ...customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux --> ready in four weeks.

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    VOIP yate call pass 6 hari left
    DISAHKAN

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    VOIP phone system 6 hari left
    DISAHKAN

    we need a phone system based on twilio platform.

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    19 bida

    Need to configure AWS WAF and Shield configuration on platform running with ELB

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    integrate bitrix with asterisk system -

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    ...“Neverwrote”, for managing virtual notes—The backend shall run on a Node.js server, and that the frontend shall be built with React. You will only need to work in the api/ directory (for the backend API) and the frontend/ directory (for the frontend interface). The NGINX server is already correctly configured, as are all of the Docker-related things

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    Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

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    No matter what the visitor types [login to view URL] or [login to view URL] or [login to view URL] i want all of the requests to be redirected to [login to view URL] currently i have managed to redirected requests that are coming from [login to view URL] and [login to view URL] to [login to view URL] but i can't find a solution for [login to view URL] to [login to view URL]

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    VOIP Project 5 hari left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    FreePBX support 5 hari left

    Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from

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    asterisk expert needed 5 hari left
    DISAHKAN

    Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]

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    Step 1: Homepage with logins Step2: Login page with details step3: VOIP or Live Stream features

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    Integrazione api voip 3 hari left
    DISAHKAN

    Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

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    Paid VOIP services app 3 hari left
    DISAHKAN

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

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    configuracion de internos y de central asterisk

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    Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem

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    ...Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call ...

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    Template and Invoice Maker 2 hari left
    DISAHKAN

    To per...to PDF for use. The server used has 3 drives. One for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already [login to view URL] is a Linux only pr...

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    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

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    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    **Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development

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    I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?

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    Project description: Migration of 200 web sites from sixteen Redhat 3.0 to four CentOS 7 servers. Scope: - 200 websites running on: o apache o html (http/https) o php o perl o tomcat o mysql o vsftp - create ftp user accounts - create and configure sites on target servers - move sites contents, ssl certs and mydsl db’s - configure the

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    ...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message

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    Hi, i would like to cooperate with an expert on FreeSwitch configuration. Just to make it straight, I don't plan to use any web interface as a plugin extension like FusionPBX etc. I want configurations from the filesystem. Currently, I just have two questions. But the person that will help me on those items i plan to hire with a monthly consultation

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    hello, im looking some one talented, who can setup 123flash chat and make android and iphone application for mobil chat. only experiment guy's please!

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    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    Asterisk & Zoho CRM 1 hari left
    DISAHKAN

    We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information

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    Project description: Migration of 200 web sites from sixteen Redhat 3.0 to four CentOS 7 servers. Scope: - 200 websites running on: o apache o html (http/https) o php o perl o tomcat o mysql o vsftp - create ftp user accounts - create and configure sites on target servers - move sites contents, ssl certs and mydsl db’s - configure the

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    ...system 53 - Insurance lead generation 54 - Inbound Call Tracking 55 - Outbound Call Tracking 56 - Call Analytics integrate Asterisk call center with CRM to have some feature bellow: - Sale can use Call feature on CRM via Asterisk. - Can use feature Click to Call? Please show me the solutions (brief, draft with your paper) to continue!...

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    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    We Need somene that have some skills over ISSABEL Platform VOIP , Asterisk and Call Center Platform . We need to implemented some news features into reports database .

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    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    asterisk ari recording Akan Tamat left
    DISAHKAN

    I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...

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    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Asterisk based Soft Switch [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Atualização do site, segue as atividades| 0.1. Conversor de tabelas para Rocket (central asterisk) | 0.2 Integração entre o Rocket (central asterisk) e PROSPECTS (area administrativa) | 0.3 Inclusao do Te Ligamos (click to call) e botao blog no menu area cliente do site 1. Blog (do 0) | 2. Integração com Rd Station | [login to view URL] de cl...

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    La RCSOft software specializzata da anni in sviluppo software gestionali, per potere integrare al suo interno un CRM richiede una interfaccia con il Centralino Voip 3CX v15 in poi, per potere effettuare delle telefonate in uscita e per potere recuperare il numero chiamante in entrata in modo da posizionarsi sul form del cliente. L'interfaccia deve

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    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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    I have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

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    We already having third party integration for voip out going calling for any phone number in world. We want to create our own solution and wanted to integrate Asterisk server with incoming and outgoing call center for our business in US. 1. Need a person who knows exactly how to do that using asterisk and what other external integration options we

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